U.S. patent number 6,167,119 [Application Number 09/006,033] was granted by the patent office on 2000-12-26 for providing enhanced services through siv and personal dial tone.
This patent grant is currently assigned to Bell Atlantic Network Services, Inc.. Invention is credited to Dale L. Bartholomew, Robert D. Farris, Alexander I. McAllister, Michael J. Strauss.
United States Patent |
6,167,119 |
Bartholomew , et
al. |
December 26, 2000 |
Providing enhanced services through SIV and personal dial tone
Abstract
An intelligent telephone network provides personalized
communication services based on subscriber prescribed double speech
signal processing of utterances of both calling and answering
parties on a subscriber line having multiple subscribers with a
single directory number. Specifically, when one of the multiple
subscribers has personalized voice mail service and a busy/no
answer call is received, the network uses speech processing of an
utterance of the calling party to identify a customer service
profile of the called party in the terminating switch. This
contains instructions inviting storage of a voice message left by
the caller. Upon the subscriber going off-hook, the customer
profile of the subscriber line is installed in the switch. The
subscriber transmits an utterance and the personalized customer
profile of the subscriber is identified by a virtual office
equipment number and the profile installed in the switch. The
subscriber may then retrieve the stored message.
Inventors: |
Bartholomew; Dale L. (Vienna,
VA), Farris; Robert D. (Sterling, VA), McAllister;
Alexander I. (Silver Spring, MD), Strauss; Michael J.
(Potomac, MD) |
Assignee: |
Bell Atlantic Network Services,
Inc. (Arlington, VA)
|
Family
ID: |
46254706 |
Appl.
No.: |
09/006,033 |
Filed: |
January 12, 1998 |
Related U.S. Patent Documents
|
|
|
|
|
|
|
Application
Number |
Filing Date |
Patent Number |
Issue Date |
|
|
828959 |
Mar 8, 1997 |
5978450 |
|
|
|
904936 |
Aug 1, 1997 |
6038305 |
|
|
|
997505 |
Dec 23, 1997 |
6101242 |
|
|
|
Current U.S.
Class: |
379/88.04;
379/201.02; 379/221.09; 379/221.12 |
Current CPC
Class: |
H04M
3/38 (20130101); H04M 3/42 (20130101); H04M
3/42017 (20130101); H04M 3/42229 (20130101); H04M
3/533 (20130101); H04Q 3/0029 (20130101); H04Q
3/0045 (20130101); H04M 3/2281 (20130101); H04M
3/382 (20130101); H04M 3/385 (20130101); H04M
3/42059 (20130101); H04M 3/42068 (20130101); H04M
3/42102 (20130101); H04M 3/4211 (20130101); H04M
3/53383 (20130101); H04M 7/06 (20130101); H04M
7/12 (20130101); H04M 2201/40 (20130101); H04M
2201/405 (20130101); H04M 2207/12 (20130101) |
Current International
Class: |
H04M
3/42 (20060101); H04M 003/42 () |
Field of
Search: |
;379/67.1,88.01,88.02,88.03,88.04,88.22,88.26,201,207 |
References Cited
[Referenced By]
U.S. Patent Documents
Primary Examiner: Weaver; Scott L.
Attorney, Agent or Firm: McDermott, Will & Emery
Parent Case Text
CROSS-REFERENCE TO RELATED APPLICATIONS
This is a continunation-in-part of U.S. patent applications Ser.
No. 08/828,959, filed Mar. 8, 1997, now U.S. Pat. No. 5,978,450;
08/904,936, filed Aug. 1, 1997, now U.S. Pat. No. 6,038,305; and
08/997,505, filed Dec. 23, 1997 , now U.S. Pat. No. 6,101,242 the
disclosures of which are incorporated entirely by reference.
Claims
What is claimed is:
1. A method comprising:
detecting a request to make a voice call from a first link to an
identified second link through a communication network including
multiple central office switching systems connected to multiple
links;
identifying one of multiple parties available via said second link
by processing signals resulting from speech transmitted via said
first link;
selecting a customer profile record corresponding to a virtual
office equipment designation assigned to said one party;
installing said customer profile record at a central office
switching system serving said second link;
pursuant to information in said customer profile record storing a
voice message in message storage associated with said second link
and said one party for retrieval by said one party.
2. A method according to claim 1 including the step of transmitting
said stored message responsive to identification by processing
signals resulting from an utterance of a party retrieving said
message.
3. A method according to claim 2 wherein said transmitting said
stored message is responsive to said customer profile record
following its reinstallation at said central office switching
system serving said second link.
4. A method according to claim 3 wherein said identification by
processing signals resulting from an utterance of a party
retrieving said message is at least partially responsive to
information contained in a customer profile record different than
said customer profile record corresponding to a virtual office
equipment designation assigned to said one party.
5. A method according to claim 4 wherein said reinstallation of
said customer profile record is responsive to said identification
by processing signals resulting from an utterance of a party
retrieving said message.
6. A method according to claim 1 wherein said storing of a voice
message is responsive to a busy or no answer condition at said
second link.
7. A method according to claim 6 wherein said storing of a voice
message is responsive to a voice prompt responsive to detection of
said busy or no answer condition at said second link.
8. A method according to claim 1 wherein said storing of said voice
message is responsive to processing of speech signals transmitted
from said first link.
9. A method according to claim 8 wherein said last named speech
comprises at least a portion of said speech transmitted via said
first link identifying said one of multiple parties available via
said second link.
10. A method according to claim 1 wherein each of said multiple
parties is assigned a personalized customer profile record
corresponding to an individual virtual office equipment
designation.
11. A method comprising:
detecting a request to make a voice call from a first link to a
second link through a switched telecommunication network including
multiple central office switching systems connected to multiple
links;
installing a first customer profile record corresponding to an
office equipment designation assigned to said second link in a
central office switching system serving said second link:
pursuant to information in said first customer profile record
establishing a speech connection between said first link and a
speech processing facility;
identifying by speech processing an utterance identifying a called
party, said utterance transmitted via said first link to said
speech processing facility;
selecting a second customer profile record corresponding to a
virtual office equipment designation assigned to said called
party;
replacing said first customer profile record with said second
customer profile record;
pursuant to information in said second customer profile record
establishing a speech connection between said first link and a
message storage associated with said second link;
storing a voice message for said called party in said message
storage.
12. A method according to claim 11 wherein said utterance
identifying said called party emanates from a party initiating said
request to make said voice call.
13. A method according to claim 11 including the step of
transmitting said stored message responsive to speech processing
identification of a party requesting said message.
14. A method according to claim 13 wherein said speech processing
is responsive at least in part to information in said second
customer profile record following reinstallation of said second
customer profile record at said central office switching system
serving said second link.
15. A method according to claim 14 wherein said reinstallation is
responsive to processing of speech of said called party.
16. A method according to claim 15 wherein the speech of said
called party which is processed is received by said processing
facility via said second link.
17. A method according to claim 11 wherein said second link is
associated with multiple customers, each of said multiple customers
being assigned a personalized customer profile record corresponding
to an individual virtual office equipment designation.
18. A method for processing a call in multilink telecommunication
network comprising the steps of:
assigning subscriber profiles to subscribers associated with a
first of said links connected to a first switching system;
at least one of said subscriber profiles including data indicative
of subscription to messaging service;
designating in a messaging system a storage capability for said at
least one subscriber profile;
identifying through speech signal processing the subscriber profile
of one of said subscribers to whom an attempt is made to initiate a
call via a second link;
processing said attempt to initiate said call in a manner based at
least in part on processing information in the subscriber profile
identified by said speech signal processing including said data
indicative of subscription to messaging service;
establishing a speech link between said second link and said
messaging system; and
storing in said designated storage capability in said messaging
system a message transmitted via said second link.
19. A method according to claim 18 wherein said subscriber profile
identified by said speech signal processing is selected from said
subscriber profiles assigned to said multiple subscribers
associated with said first of said links by a virtual equipment
designation.
20. A method according to claim 18 including the steps of assigning
virtual office equipment designations to each of said subscriber
profiles, and selecting from said subscriber profiles said
subscriber profile identified by said speech signal processing by
its virtual equipment designation.
21. A method according to claim 20 including the steps of assigning
to said first link an office equipment designation designating
office equipment in a switching system in said multilink
telecommunication network, which switching system serves said first
link, said office equipment designation identifying a further
customer profile, and identifying through speech signal processing
the subscriber profile of said one of said subscribers to whom an
attempt is made to initiate a call via said second link pursuant at
least in part to information in said further customer profile.
22. A method according to claim 21 including the steps of
installing said further customer profile in said switching system
responsive to detecting said attempt to initiate a call via said
second link, and replacing said further customer profile in said
switching system with said customer profile identified through
speech signal processing.
23. A method for processing a call in a switched telecommunication
network including multiple central office switching systems
connected to multiple links comprising the steps of:
assigning a customer profile to a first of said multiple links
corresponding to office equipment in a central office switching
system serving said first link;
assigning additional subscriber profiles to multiple subscribers
associated with said first link;
at least one of said additional subscriber profiles including data
indicative of subscription to messaging service;
designating in a messaging system a storage capability for said at
least one additional subscriber profile;
installing in said switching system serving said first link said
customer profile assigned to said first link;
pursuant at least in part to information in said installed customer
profile identifying through speech signal processing the subscriber
profile of one of said subscribers to whom an attempt is made to
initiate a call via a second link;
substituting for said installed customer profile said customer
profile identified through speech signal processing;
processing said attempt to initiate said call in a manner based at
least in part on processing information in the substituted
subscriber profile identified by said speech signal processing
including said data indicative of subscription to messaging
service;
establishing a speech link between said second link and said
messaging system; and
storing in said designated storage capability in said messaging
system a message transmitted via said second link.
24. A method according to claim 23 wherein said additional customer
profiles are designated by virtual office equipment designations
associated with said office equipment for said first link.
25. A method according to claim 24 wherein said speech signal
processing comprises recognizing the utterance of a name.
26. A method according to claim 25 wherein said utterance of said
name emanates from the party attempting to initiate said call.
27. A method according to claim 24 including the step of
transmitting said stored message responsive to speech signal
processing identification of a party retrieving said message.
28. A method according to claim 27 wherein said speech signal
processing identification of a party retrieving said message is
responsive at least in part to information in said customer profile
corresponding to said office equipment for said first link.
29. A method according to claim 28 wherein said step of
transmitting said stored message responsive to speech signal
processing identification of a party retrieving said message is
responsive at least in part to information in said customer profile
identified by a virtual office equipment designation assigned to
said party retrieving said message.
30. A method comprising:
detecting a request to make a voice call from a first link to a
second link through a communication network including multiple
central office switching systems connected to multiple links;
installing in the central office switching system serving said
second link a first service profile corresponding to said second
link;
pursuant to said first service profile identifying one of multiple
parties available via said second link by processing signals
resulting from speech transmitted via said first link;
responsive to said identification installing in the central office
switching system serving said second link a second service profile
corresponding to said one party;
pursuant at least partially to information in said second service
profile storing a voice message from said first link in message
storage associated with said second link and with said one party
for retrieval by said one party.
31. A method according to claim 30 wherein said first service
profile corresponds to designated office equipment in said central
office switching system serving said second link.
32. A method according to claim 31 wherein said second service
profile corresponds to designated virtual office equipment in said
central office switching system serving said second link.
33. A method according to claim 32 wherein said second service
profile is substituted for said first service profile in said
central office switching system serving said second link.
34. A method according to claim 30 including the steps of:
detecting a request for service over said second link;
installing said first service profile in said central office
switching system serving said second link;
responsive to a voice signal over said second link substituting
said second service profile for said first service profile in said
central office switching system serving said second link;
pursuant at least partially to information in said second service
profile transmitting said stored voice message via said second
link.
35. A method according to claim 33 including the steps of:
detecting a request for service over said second link;
installing said first service profile in said central office
switching system serving said second link;
responsive to a voice signal over said second link substituting
said second service profile for said first service profile in said
central office switching system serving said second link;
pursuant at least partially to information in said second service
profile transmitting said stored voice message via said second
link.
36. A communication network comprising:
multiple central office switching systems connected to multiple
links, said links including a first and a second link, said second
link having multiple subscribers associated therewith;
a control network connected to said central office switching
systems;
a speech signal processing facility; and
a message storage facility;
wherein responsive to initiation of a request to establish a
connection from said first to said second links, a connection is
established to said speech signal processing facility, the identity
of one of said subscribers associated with said second link is
established via said speech signal processing facility, a virtual
office equipment designation identifying a service profile of said
one subscriber is associated with the switching system to which
said second link is connected, pursuant at least partially to
information in said service profile storing a voice message from
said first link in said message storage facility for retrieval by
said identified subscriber.
37. A communication network comprising:
multiple central office switching systems connected to multiple
links, said links including a first and a second link, said second
link having multiple subscribers associated therewith;
a speech signal processing facility; and
a message storage facility;
a signaling and control network connected to said central office
switching systems, said speech signal processing facility; and said
message storage facility;
wherein responsive to initiation of a request to establish a
connection from said first to said second links, a connection is
established to said speech signal processing facility, the identity
of one of said subscribers associated with said second link is
established, a virtual office equipment designation identifying a
service profile of said one subscriber is associated with the
switching system to which said second link is connected, pursuant
at least partially to information in said service profile storing a
voice message from said first link in said message storage facility
for retrieval by said identified subscriber.
38. A communication network comprising:
multiple central office switching systems connected to multiple
links, said links including a first and a second link, said second
link having multiple subscribers associated therewith;
a speech signal processing facility;
a message storage facility;
a signaling and control network connected to said central office
switching systems, said speech signal processing facility; and said
message storage facility;
wherein responsive to initiation of a request to establish a
connection from said first to said second links, a first service
profile corresponding to said second link is installed in the
central office switching system serving said second link and
pursuant at least partially to information in said first service
profile connection is established to said speech signal processing
facility, the identity of one of said subscribers associated with
said second link is established, a second service profile
corresponding to said one subscriber is installed in said central
office switching system serving said second link, pursuant at least
partially to information in said second service profile a voice
message from said first link is stored in message storage
associated with said second link for retrieval by said one
party.
39. A system according to claim 38 wherein said switching systems
are connected in a telecommunications network.
40. A system according to claim 39 wherein responsive to detecting
a request for service over said second link subsequent to storage
of said message, said first service profile in installed in said
central office switching system serving said second link,
responsive to an utterance over said second link, said second
service profile is substituted for said first service profile in
said central office switching system serving said second link, and
pursuant at least partially to information in said second service
profile said stored voice message is transmitted via said second
link.
41. A method comprising:
detecting a request to make a voice call from a first link to a
second link through a communication network including multiple
central office switching systems connected to multiple links;
installing in the central office switching system serving said
second link a service profile generic to said second link;
pursuant to said generic service profile identifying one of
multiple parties available via said second link by processing
signals resulting from speech transmitted via said first link;
responsive to said identification installing in the central office
switching system serving said second link a service profile
specific said one party;
pursuant at least partially to information in said specific service
profile storing a voice message from said first link in message
storage associated with said second link and with said one party
for retrieval by said one party.
Description
TECHNICAL FIELD
The present invention relates to personalized telecommunications
service, preferably offered through an intelligent telephone
network. In particular, the present invention relates to the
identification of one or both calling and answering speakers to
control processing of the communication. Enhanced services are
provided on a personalized basis to multiple subscribers using the
same line to terminal equipment.
Acronyms
The written description uses a large number of acronyms to refer to
various services, messages and system components. Although
generally known, use of several of these acronyms is not strictly
standardized in the art. For purposes of this discussion, acronyms
therefore will be defined as follows:
Address Complete Message (ACM)
Advanced Intelligent Network (AIN)
ANswer Message (ANM)
Automatic Number Identification (ANI)
Call Processing Record (CPR)
Central Office (CO)
Common Channel Interoffice Signalling (CCIS)
Data and Reporting System (DRS)
Destination Point Code (DPC)
Generic Data Interface (GDI)
Initial Address Message (IAM)
Integrated Service Control Point (ISCP)
Integrated Services Digital Network (ISDN)
ISDN User Part (ISDN-UP)
Intelligent Peripheral (IP)
Line Identification Data Base (LIDB)
Multi-Services Application Platform (MSAP)
Office Equipment (OE)
Origination Point Code (OPC)
Personal Communications Service (PCS)
Plain Old Telephone Service (POTS)
Point in Call (PIC)
Personal Identification Number (PIN)
Primary Rate Interface (PRI)
Public Switched Telephone Network (PSTN)
Service Control Point (SCP)
Service Creation Environment (SCE)
Service Management System (SMS)
Service Switching Point (SSP)
Signaling System 7 (SS7)
Signaling Point (SP)
Signaling Transfer Point (STP)
Simplified Message Desk Interface (SMDI)
Speaker Identification/Verification (SIV)
Terminating Attempt Trigger (TAT)
Time Slot Interchange (TSI)
Traffic Service Position System (TSPS)
Transaction Capabilities Applications Part (TCAP)
Transmission Control Protocol/Internet Protocol (TCP/IP)
BACKGROUND ART
Today the public switched telephone network (PSTN) and other
telephone networks such as cellular systems provide most telephone
services based on number identification of the telephone set or
line that each party uses. Services are personalized only to the
extent that a party uses the same line and/or instrument. For
example, a person typically has one set of service features and
billing options available via a telephone on the person's desk at
the office, another set of service features and billing options
available via the telephone line to their home and perhaps a third
set of service features and billing options available via a
wireless telephone (e.g. cellular or personal communications
service (PCS)). The networks process calls to and from each of
these different subscriber telephones based on a separate telephone
number. Also, a caller may use personalized billing options by
using a calling card, but often the input operations for calling
card service are overly complex. With the exception of calling card
billing, each person using a particular telephone typically can
only access those service features and billing options associated
with the particular line or telephone instrument.
The proliferation of services causes subscribers inconvenience. For
example, circumstances arise in which a subscriber may want a
feature or billing option normally associated with one line or
instrument, such as the office telephone, when they are in fact
using a different line or instrument such as their home or PCS
telephone. Alternatively, two or more persons using one telephone
or line often want different sets of service options. Also, the
extreme increase in demand for telephone services is rapidly
exhausting the capacity of the network, particularly in terms of
the telephone numbers available under the current numbering
plan.
A number of specific solutions have been proposed for individual
problems, such as work at home and/or transfer of service to new
locations) as an individual travels. However, each of these
solutions is limited or creates its own new problems.
For example, U.S. Pat. No. 4,313,035 to Jordan et al. discloses a
method of using an intelligent network to provide a `follow-me`
type service through multiple exchanges of the switched telephone
network using an AIN type of telephone system architecture. Each
subscriber to the locator service has a unique person locator
telephone number. To access the system to update data in a service
control database, the subscriber dials 0700 and his unique person
locator telephone number. The telephone switching office routes the
call to a traffic service position system (TSPS) which prompts the
caller (e.g. provides an additional dial tone) and receives further
digits from the subscriber. The subscriber inputs a three digit
access code, indicating the type of update call, and a four digit
personal identification number. If calling from the remote station
to which the subscriber wishes his calls routed, the local
switching office forwards the line identification number of that
station to the TSPS. The TSPS forwards the dialed information and
the line identification to the data base for updating the
particular subscriber's location record. A caller wishing to reach
the subscriber dials the subscriber's unique person locator number.
A telephone switching office sends the dialed number to the central
database. The database retrieves the stored completion number for
the called subscriber and forwards that number back to the
switching office to complete the call.
The Jordan et al. approach allows calls to follow the subscriber to
each new location, but the subscriber must have a unique telephone
number for this service. Each station that receives a call also
must have a unique telephone number. As such, the Jordan et al.
approach actually exacerbates the shortage of telephone numbers.
Also, Jordan et al. rely on subscriber input of identification
numbers. Subscribers often find this inconvenient, and this
technique is often prone to number entry errors.
U.S. Pat. No. 4,899,373 to Lee et al. discloses a system for
providing special telephone services to a customer on a personal
basis, when the customer is away from his or her home base or
office. The personalized services are provided in a multiple
exchange office environment, using a central database for feature
control. The nationally accessible central database system stores
feature data in association with personal identification numbers. A
subscriber wishing to use his personalized features while away from
home base dials a special code and presents the personal
identification number. The exchange transmits a query to the
central database, and the corresponding feature data is retrieved
from the database. The database forwards the feature data to the
exchange, and the exchange stores the received feature data in
association with the station from which the request was initiated.
Subsequently, the exchange accesses the downloaded feature data to
provide telephone service corresponding to the subscriber's
personalized telephone features via the station the subscriber is
currently operating from. A temporary office arrangement may be
established in which the personalized features will be immediately
available on incoming and outgoing calls for a period of time
specified by the subscriber.
U.S. Pat. No. 5,206,899 to Gupta et al. pertains to a system
wherein a subscriber can assign desired characteristics to any
"target station" which is an active telephone accessible to a
telecommunications network. A call thereafter that originates from
the target station can use customized features, such as account
code dialing and corporate billing arrangements. Initially, a
service profile is created and stored for each subscriber and
contains information describing desired features and billing
options. The characteristics of a particular target station are
changed by an activation process that can be initiated from any
location. Automatic number identification (ANI) information
associated with the target station is entered into an ANI trigger
table in an intelligent switch, and the service profile is loaded
into a database. When a call originates from the target station,
information in the database is applied to the switch to provide the
desired characteristics. An example of one of the features is when
an employee of company X wishes to make business related calls from
his/her telephone, the call has the characteristics of a call made
from the office by a special billing arrangement.
Like Jordan, the Lee et al. and Gupta et al. systems depend on a
dialed number entry by the subscriber to activate the service.
Also, the Lee et al. and Gupta et al. systems do not provide a
simple manner for more than one subscriber to obtain personalized
service over the same telephone line. In Lee et al., during the
period when the switch stores the roaming subscriber's profile in
association with the line, all calls are processed based on that
one profile. Similarly, in Gupta et al., while the ANI trigger is
set against the line, all outgoing calls cause database access and
use of the subscriber's profile in the database. There is no way to
fall back on the normal profile for that line unless and until the
service for the roaming subscriber is cancelled with respect to
that one line.
U.S. Pat. No. 5,247,571 to Kay et al. discloses an Area Wide
Centrex service provided by an advanced intelligent telephone
network. The service provides centrex features, such as extension
dialing, to multiple locations. The Kay et al. Patent also suggests
a Work-at-Home feature. This feature allows the home telephone line
to selectively operate as a residential line or as a Centrex
business line, on a call-by-call basis. For a business call, the
user would preface each call with an access indicator to identify a
business call. When an outgoing call from the home line lacks the
access indicator, the network processes the call as a standard
residential call.
The Work-at-Home feature in the Kay et al. system requires only
dialing of a code before each outgoing business call. However, the
Kay et al. approach requires that the business profile is stored in
association with the home line before the subscriber makes the
call. The subscriber can use the Centrex billing and service
features from the business account only from a home telephone
previously associated with the business line. The subscriber can
not use the billing and service features from the business account
from any randomly selected telephone. Also, from the home line, a
person can either use the normal residential profile service or the
pre-defined business profile service. There is insufficient
flexibility to enable a wider range of services for multiple
subscribers through the one line.
U.S. Pat. No. 5,422,936 to Douglas J. Atwell, issued Jun. 6, 1995,
describes an Enhanced Message Service Indication. For a number of
years, telephone companies have been providing a service which
assigned two or more directory numbers per line and corresponding
distinctive ringing signals. One of the telephone switch vendors
refers to this feature as "Multiple Directory Numbers per Line" or
"MDNL." This patent provides a system for providing voice mail
service in a MDNL situation. The system is effective in serving its
intended purpose but assumes the assignment of one directory or
telephone number for each subscriber or service. As previously
stated the current demand for telephone services is rapidly
exhausting the capacity of the network, particularly in view of the
telephone numbers available under the current numbering plan.
An increasingly popular telephone services is caller identification
or `caller ID`. The telephone network identifies the telephone
number associated with the line or instrument used by the calling
party and supplies the number and or the name to a display device
at the called customer's premises.
Subscribers having ISDN service receive caller ID data, for display
at the time of an incoming call, in the form of a data message
which the end office switch transmits over the D-channel. For
analog telephone customers, however, existing caller ID utilizes
in-band transmission technology similar to that described in U.S.
Pat. Nos. 4,582,956 and 4,551,581 to Doughty. In such a system, the
end office switch connected to the called party's line transmits
directory number data for the calling party's telephone line as
frequency shift keyed (FSK) data inserted in the silent interval
between ringing signal pulses applied to the called party's line.
The receiving apparatus includes a line interface unit, a
converter, a control circuit and a display unit. A frequency shift
keyed (FSK) signal representing the special service information is
filtered from the ringing signals by the line interface unit. The
converter detects the FSK signal and demodulates the special
service information from the FSK signal. Following detection of the
FSK signal, the control circuit receives and stores the special
service information. The stored information is periodically sent to
the display unit to begin exhibiting thereof during the silent
interval before the next ringing signal.
The local telephone exchange carriers have recently begun offering
an enhanced form of caller ID, sometimes referred to as `Caller ID
Deluxe` service. This enhanced service utilizes AIN type call
processing to access a Line Information Database (LIDB) to
translate the calling party's directory number into name data. The
end office switch forwards the name data and the normal caller ID
telephone number as FSK encoded data inserted in the silent
intervals between ringing signals.
The LIDB database includes a single listing for each telephone line
and translates each number into a single name, typically the name
of the party identified as the customer or subscriber for billing
purposes. In fact, the LIDB database provides this single
translation even for calls from one line having multiple telephone
numbers. Consider an example in which a family has one line with
two numbers and a distinctive ringing service. The first number is
used for the family as a whole, and the second number is used for a
teenage son or daughter. The distinctive ringing allows people in
the household to know whether or not each call is for the teenager.
On outgoing calls, however, the end office switch always identifies
the line by the primary number (the family's number), and the LIDB
database always provides the name of the billing subscriber, e.g.
the father's name. As a result, when the teenager calls a friend,
the friend will receive the main number and possibly the father's
name. If the friend calls back using the information from his
caller ID display terminal, the friend calls the family's main
number, not the teenager's number.
Also, the above discussed examples of prior suggestions to
customize services have not adapted the caller identification to
correspond to the actual party using the telephone on the outgoing
call. For example, in a system like that of Lee, Gupta or Kay, the
caller might use features and billing options associated with her
personalized or work service, but any such calls would produce a
caller ID display identifying the number of the station from which
she originated the call. If the called party subscribed to the name
type enhanced caller ID, the network would provide a name
associated with that telephone number, not the name of the actual
calling party.
U.S. Pat. Nos. 4,961,217 and 4,759,056 disclose a card based system
for providing personalized features, including caller name display.
Each user has a "portable memory device" in the form of an
identification card bearing personal information including
identification information. When initiating a call, the user
inserts the card in the calling station, and information from the
card is transmitted to the central switching system. In one
embodiment, the switching system translates the identification
information from the card to produce a textual representation of
the calling party's name and transmits that information to a called
terminal for display. Although this system does provide a name
display identifying the actual called party, the system requires
the use of the identification card and specialized calling
terminals for reading the information from the cards.
Another enhanced service which has become extremely popular is so
called Voice Mail service. Voice mail is a service which may be
considered a custom calling service and normally includes in its
operation the use of call forwarding. Voice mail has become
commonplace not only in business usage but also on an individual
telephone service subscriber basis through Centrex service from a
central office. A voice mail system is a specialized computer that
stores messages in digital form on a fixed disk. The voice is
generally digitized, usually at a much slower rate than the 64 Kb/s
signal the central office uses in its switching network. The
digitized voice is compressed and stored on a hard disk that
maintains the voice mail operating system, system prompts, and
greetings, and the messages themselves. A processor controls the
compressing, storing, retrieving, forwarding and purging of files.
A form of early systems is described in Matthews et al. U.S. Pat.
No. 4,371,752 (hereinafter the Matthews '752 Patent), issued in
February, 1983, and several related patents. U.S. Pat. No.
4,585,906 (hereinafter the Matthews '906 Patent), issued Apr. 29,
1986 to Gordon H. Matthews et al. The Matthews '906 Patent is a
continuation-in-part of the Matthews '752 Patent. U.S. Pat. No.
4,602,129 (hereinafter the Matthews '129 Patent), issued Jul. 22,
1986 to Gordon H. Matthews et al. The Matthews '129 Patent is a
continuation-in-part of the '752 Matthews Patent.
The three Matthews Patents each describe a voice mailbox type
system using digital storage and programmed control to offer a wide
variety of message storage, forwarding and delivery type
services.
U.S. Pat. No. 4,625,081, issued Nov. 25, 1986, to Lawrence A.
Lotito, et al. This patent describes an automated telephone voice
service system which provides automatic recording and editing of
voice messages as well as forwarding of recorded voice messages to
other accounts and telephone numbers with or without operator
assistance.
In all of the foregoing systems voice mail is provided to a single
subscriber premises line or, as in the Atwell Patent, to a single
subscriber number. A need still exists for an effective and user
friendly system for providing personalized calling service
features, including actual subscriber identification for voice mail
purposes. In particular a need exists for a system for providing
personalized features which would facilitate a degree of call
control permitting the accomplishment of new functions, including
enhanced voice mail and voice mail notification, and which would
improve the handling of functions which are now subject to being
accomplished only in cumbersome and inconvenient fashions.
DISCLOSURE OF THE INVENTION
The present invention addresses the above noted problems and
provides advances over the existing technology by personalizing
telecommunication services based on a speech authenticated
identification of the not only of the actual subscriber but also of
the speakers at both ends of the communication. Offices of a
communication network utilize profile data associated with
identified persons, rather than profile data associated with a
particular telephone number or a particular communication link. In
many of the preferred service applications, the network uses a
virtual office equipment number assigned to a speaker's profile
data to retrieve the data for providing a specific service,
reducing or eliminating the need for assignment of additional
telephone numbers. The network also provides responding party
identification information which is used to determine at least a
portion of the processing of the particular call.
Thus, in one aspect the present invention relates to a method of
providing service through a communication network. A request to
make a call from a predetermined link through the network is
detected. The next step in the method is receiving and processing
speech signals from a person via the predetermined link. The
processing identifies the person making the call as one of a number
of subscribers or persons designated as users of services offered
through the communication network. An instruction is sent to a
switching office of the network instructing that office to utilize
profile data corresponding to the identified subscriber or user for
processing of the call. Preferably the profile data is selected at
least partially through the use of a virtual office equipment
number. This method includes identifying one party to a requested
communication service, for example the party making an outgoing
call, as one of a plurality of subscribers or designated users.
Using a virtual office equipment number, assigned to the identified
one user, corresponding profile data is retrieved from storage. A
communication network provides the requested communication service
over a communication link, based at least in part on the retrieved
profile data. As part of the service, a portion of the retrieved
profile data is used to direct processing which provides
identification of a person responding to the call over another link
of the communication network.
Other aspects of the invention relate to a communication network
implementing the personalized services, including dual caller and
responder specific identification. The system and methodology
comprehended by the invention is applicable to both outgoing as
well as incoming calls. The preferred implementation of the
communication network is an intelligent implementation of a public
switched telephone network. The preferred network includes a number
of central office switches interconnected by trunk circuits and
servicing a substantial number of telephone links. The intelligent
network also includes a service control point storing a database of
records used in controlling services provided through the central
offices. A first signaling network carries signaling messages
between the offices as well as signaling messages between the
offices and the service control point. A multifunction intelligent
peripheral is provided and also may exchange signaling information
with the service control point, preferably over a second signaling
network.
Another aspect of the invention relates to an improved central
office switching system capable of processing a call using profile
information selected in response to a virtual equipment number. An
office equipment number is `virtual` where it is assigned to an
individual subscriber, instead of to specific network equipment
such as a line termination or a specific station.
The switching system includes interface modules coupled to the
communication links and a switch providing selective communication
connections between the interface modules. An administrative module
controls connections provided by the switch. The administrative
module includes mass storage containing subscriber profiles, a
processor for providing control instructions to the switch, and a
signaling interface for signaling communication with at least one
external network node. In response to a virtual office equipment
number received via the signaling interface, e.g. from a separate
peripheral platform as discussed above, the processor retrieves a
subscriber profile corresponding to the virtual office equipment
number from the mass storage. The processor uses the retrieved
profile to process a selective connection through the switch
between two of the interface modules.
Advantages of the personal dial tone service should be readily
apparent to those skilled in the telecommunications art. For
example, in the shared line application, several subscribers can
share a single line or communication link as well as a single
telephone number. Outgoing call features, however, are personalized
to each subscriber. For example, the network can provide each user
a different level of service which, according to a preferred
embodiment of the invention, may impose restrictions on that user.
In addition, the network may direct the performance of a variety of
functions both within and without the network. These functions
preferably include the identification of the second party to the
communication, and the specific nature of the functions are at
least in part determined by that identification. The service uses
speech based identification.
Additional objects, advantages and novel features of the invention
will be set forth in part in the description which follows, and in
part will become apparent to those skilled in the art upon
examination of the following or may be learned by practice of the
invention. The objects and advantages of the invention may be
realized and attained by means of the instrumentalities and
combinations particularly pointed out in the appended claims.
BRIEF DESCRIPTION OF DRAWINGS
The drawing figures depict the present invention by way of example,
not by way of limitations. In the figures, like reference numerals
refer to the same or similar elements.
FIG. 1 is a simplified block diagram of an intelligent telephone
network that may be used to offer the personalized service of the
present invention.
FIG. 2 is a simplified block diagram illustrating the significant
functional components of an SSP type central office switching
system used in the network of FIG. 1.
FIG. 3 is a simplified block diagram illustrating the significant
functional components of an Intelligent Peripheral (IP) used in the
network of FIG. 1.
FIG. 4A is a combination signal flow and process flow diagram
useful in understanding a specific example of call processing for
providing an illustrative personalized service over a shared use
line.
FIG. 4B is a combination signal flow and process flow diagram
useful in understanding one embodiment of the processing for
providing the identity of the actual caller to the destination
display as caller ID information.
FIG. 4C is a combination signal flow and process flow diagram
useful in understanding another embodiment of the processing for
providing the identity of the actual caller to the destination
display as caller ID information.
FIG. 5 is a combination signal flow and process flow diagram useful
in understanding a specific example of call processing for
providing an illustrative personalized service on a dial-up, per
call basis.
FIG. 6 is a block diagram depicting an example of one voice mail
system suitable for use pursuant to one preferred embodiment of the
invention.
BEST MODE FOR CARRYING OUT THE INVENTION
In response to each of several types of service requests, the
personalized service of the present invention initially identifies
the individual subscriber or user, preferably using a speaker
identification/verification procedure. The system then retrieves
profile information corresponding to the identified subscriber or
user. The communication network processes one or more calls to or
from an identified communication link using the individual user's
profile data. On an outgoing telephone call from the subscriber or
user, for example, the service request may be an off-hook signal,
and the network may provide `dial-tone` type telephone services
based on the retrieved profile information. In this example, the
network may provide a dial tone signal or a customized prompt and
then permit the caller to out-dial a call. Caller identification,
calling features and/or additional identification of the responding
party functions apply based on the profile information. The network
also provides personalized services on incoming calls based on the
identity of the calling party and on data contained in the
individual profile of the answering user.
The personalized service may utilize a variety of different
networks. For example, the service may be adaptable to Internet
based voice communications. The preferred embodiments utilize
various implementations of modern telephone networks. To understand
the invention, it may be helpful first to consider the architecture
and operation of an advanced intelligent network (AIN) type
implementation of a public switched telephone network.
FIG. 1 provides a simplified illustration of the preferred
intelligent telephone network for implementing the personal dial
tone service in accord with the present invention. As shown, the
telephone network includes a switched traffic network and a common
channel signaling network carrying the control signaling messages
for the switched telephone traffic network. In this implementation,
the system further includes a secondary signaling network.
The telephone or traffic network (operated by a combination of
local carriers and interexchange carriers) includes a number of end
office and tandem office type central office switching systems 11.
FIG. 1 shows a number of subscriber stations, depicted as
telephones 1, connected to a series of central office switches
11.sub.1 to 11.sub.N. In the preferred implementation, the
connections to the central office switches 11 utilize telephone
lines, and the switches are telephone type switches for providing
landline communication. However, it should be recognized that other
communication links and other types of switches could be used.
Trunk circuits TR carry communication traffic between the central
office switches 11.
Each end office type central office switch, such as 11.sub.1 and
11.sub.N, provides switched telephone connections to and from local
communication lines or other subscriber links coupled to end users
stations or telephone sets 1. For example, the central office
11.sub.1 serves as an end office to provide switched telephone
connections to and from local communication lines coupled to end
users telephone station sets, such as telephone 1.sub.A, whereas
the central office 11.sub.N serves as an end office to provide
switched telephone connections to and from local communication
lines coupled to end users telephone station sets, such as
telephone 1.sub.B.
The typical telephone network also includes one or more tandem
switching offices such as office 11.sub.T, providing trunk
connections between end offices. As such, the traffic network
consists of local communication links and a series of switching
offices interconnected by voice grade trunks, only two examples of
which are shown at TR in FIG. 1. One set of trunks TR might
interconnect the first end office 11.sub.1 to the tandem office
11.sub.T, whereas another set of trunks TR might interconnect the
tandem office 11.sub.T to another end office 11.sub.N. Other trunks
might directly connect end offices. Although not shown, many
offices serve as both end offices and tandem offices for providing
different call connections.
FIG. 1 shows connections to the stations 1 via lines, and typically
these links are telephone lines (e.g. POTS or ISDN). It will be
apparent to those skilled in the art, however, that these links may
be other types of communication links, such as wireless links. At
least some of the stations have caller ID capability. If the line
is an ISDN line, the station may incorporate a display for visually
presenting the caller ID information and other signaling related
messages. If the link is a typical analog telephone line, the
customer premises equipment includes a caller ID terminal, one
example of which is shown at 5.sub.B. The terminal 5.sub.B displays
at least telephone numbers and preferably displays alphanumeric
information to enable displays of callers names.
Although shown as telephones in FIG. 1, the terminal devices or
stations 1 can comprise any communication device compatible with
the local communication link. Where the link is a standard voice
grade telephone line, for example, the terminals could include
facsimile devices, modems etc. The processing in accord with the
invention, however, relies on identification of the subscriber,
preferably by voice based recognition. For this purpose, the
terminals preferably include two-way voice communication
elements.
The lines and trunks through the central offices 11 carry the
communication traffic of the telephone network. The preferred
telephone network, however, also includes a common channel
interoffice signaling (CCIS) network carrying a variety of
signaling messages, principally relating to control of processing
of calls through the traffic portion of the network. The CCIS
network includes packet data links (shown as dotted lines)
connected to appropriately equipped central office switching
systems such as offices 11 and a plurality of packet switches,
termed Signaling Transfer Points (STPs) 15. To provide redundancy
and thus a high degree of reliability, the STPs 15 typically are
implemented as mated pairs of STPs. The CCIS network of the
telephone system operates in accord with an accepted signaling
protocol standard, preferably Signaling System 7 (SS7).
In the preferred embodiment shown in FIG. 1, each central office 11
has at least minimal SS7 signaling capability, which is
conventionally referred to as a signaling point (SP) in reference
to the SS7 network. As such, the offices can exchanges messages
relating to call set-up and tear-down, typically in ISDN-UP format.
At least some, and preferably all, of the central office switches
11 are programmed to recognize identified events or points in call
(PICs) as advanced intelligent network (AIN) type service triggers.
In response to a PIC or trigger, a central office 11 initiates a
query through the CCIS signaling network to a control node to
either a Service Control Point (SCP) 19 or to a database system,
such as a Line Identification Database (LIDB) 21. The SCP 19
provides instructions relating to AIN type services. The LIDB 21
provides subscriber account related information, for calling card
billing services or for subscriber name display purposes in an
enhanced caller ID application. Those central office switching
systems having full AIN trigger and query capability for
communication with the SCP and/or the LIDB are referred to as
Service Switching Points (SSPs).
The central office switches 11 typically consist of programmable
digital switches with CCIS communications capabilities. One example
of such a switch is a 5ESS type switch manufactured by AT&T;
but other vendors, such as Northern Telecom and Seimens,
manufacture comparable digital switches which could serve as the
SSPs and SPs. The SSP type implementation of such switches differs
from the SP type implementation of such switches in that the SSP
switch includes additional software to recognize the full set of
AIN triggers and launch appropriate queries. A specific example of
an SSP capable switch is discussed in detail later, with regard to
FIG. 2.
One key feature of the present invention is that the program
controlled switch accepts instructions to load profiles and/or
receives profiles over a signaling link. In most cases, these
profiles are identified by virtual office equipment numbers. The
profiles include a range of information relating to subscribers
services, such as service features, classes of service, individual
billing options, and according to a preferred feature of the
invention, information relating to restrictions applied to
individual users, as well as the performance of functions related
to that user.
The above described data signalling network between the SSP type
central offices 11 and the SCP 19 is preferred, but other
signalling networks could be used. For example, instead of the
packet switched type links through one or more STP's, a number of
central office switches, an SCP and any other signaling nodes could
be linked for data communication by a token ring network. Also, the
SSP capability may not always be available at the local office
level, and several other implementations might be used to provide
the requisite SSP capability. For example, none of the end office
switches may have SSP functionality. Instead, each end office would
connect through a trunk to a tandem office which has the SSP
capability. The SSP tandem then communicates with the SCP via an
SS7 type CCIS link, as in the implementation described above. The
SSP capable tandem switches are digital switches, such as the 5ESS
switch from AT&T; and the non-SSP type end offices might be 1A
analog type switches.
The SCP 19 may be a general purpose computer storing a database of
call processing information. In the preferred implementation, the
SCP 19 actually is an Integrated Service Control Point (ISCP)
developed by Bell Atlantic and Bell Communications Research. The
ISCP is an integrated system. Among other system components, the
ISCP includes a Service Management System (SMS), a Data and
Reporting System (DRS) and the actual database also referred to as
a Service Control Point (SCP). In this implementation, the SCP
maintains a Multi-Services Application Platform (MSAP) database
which contains call processing records (CPRs) for processing of
calls to and from various subscribers. The ISCP also typically
includes a terminal subsystem referred to as a Service Creation
Environment or SCE for programming the MSAP database in the SCP for
the services subscribed to by each individual customer.
The components of the ISCP are connected by an internal, high-speed
data network, such as a token ring network. The internal data
network also typically connects to a number of interfaces for
communication with external data systems, e.g. for provisioning and
maintenance. In the preferred embodiment, one of these interfaces
provides communications to and from the SCP 19 via a packet
switched data network, such as the TCP/IP network 27.
The SCP may be implemented in a variety of other ways. The SCP may
be a general purpose computer running a database application and
may be associated with one of the switches. Another alternative is
to implement a database of CPRs or the like within an STP (see e.g.
Farris et al. U.S. Pat. No. 5,586,177).
The LIDB database 21 is a general purpose computer system having a
signalling link interface or connection to a pair of STPs 15. The
computer runs a database program to maintain a database of
information relating to customer accounts and identifications. For
example, a subscriber's entry in the LIDB database might include
the subscriber's telephone number, a personal identification number
for credit card billing purposes, and the subscriber's name and
address.
The preferred telephone network also includes one or more
intelligent peripherals (IPs) 23 to provide enhanced announcement
and digit collection capabilities and speech recognition. The IP 23
is essentially similar to that disclosed in commonly assigned U.S.
U.S. Pat. No. 5,572,583 to Wheeler, Jr. et al. entitled "Advanced
Intelligent Network with Intelligent Peripherals Interfaced to the
Integrated Services Control Point," and the disclosure of the
network and operation of the IP disclosed from that Patent is
incorporated herein in its entirety by reference.
The IP 23 may connect to one or more central offices 11. The
connections transport both communication traffic and signaling. The
connection between a central office 11 and the IP 23 may use a
combination of a T1 and a Simplified Message Desk Interface (SMDI)
link, but preferably this connection utilizes a primary rate
interface (PRI) type ISDN link. Each such connection provides
digital transport for a number of two-way voice grade type
telephone communications and a channel transporting signaling data
messages in both directions between the switch and the IP.
As discussed more later, there are certain circumstances in which
the SCP 19 communicates with the IP 23. These communications could
utilize an 1129 protocol and go through an SSP type central office
11 and the SS7 network. However, in the preferred embodiment of
FIG. 1, the IP 23 and the SCP 19 communicate with each other via a
separate second signaling network 27. These communications through
network 27 between the IP and the SCP may utilize an 1129+ protocol
or a generic data interface (GDI) protocol as discussed in the
above incorporated Patent to Wheeler, Jr. et al.
The IP 23 can provide a wide range of call processing functions,
such as message playback and digit collection. In the preferred
system, the IP also performs speaker identification/verification
(SIV) on audio signals received from users. Specifically, the IP 23
used for the personalized service includes a voice authentication
module to perform the necessary speaker identification/verification
function. The IP 23 also includes storage for subscriber specific
template or voice feature information, for use in identifying and
authenticating subscribers based on speech.
In the simplest form, the IP 23 serving a subscriber's local area
stores the templates and performs the speaker
identification/verification. However, in a system serving a large
geographic area and providing personal dial tone to a large,
roaming subscriber base, the templates may be transferred between
SCP/IP pairs, to allow an IP near a subscriber's current location
to perform the speaker identification/verification on a particular
call. For example, if a remote IP 23.sub.R required a template for
a subscriber from the region served by the IP 23, the remote IP
23.sub.R would transmit a template request message through the
network 27 to the IP 23. The IP 23 would transmit the requested
template back through the network 27 to the remote IP 23.sub.R.
In a network such as shown in FIG. 1, routing typically is based on
dialed digit information, profile information regarding the link or
station used by the calling party and profile information regarding
a line or station in some way associated with the dialed digits.
Each exchange is identified by one or more three digit codes. Each
such code corresponds to the NXX digits of an NXX-XXXX (seven
digit) telephone number or the three digits following the area code
digits (NPA) in a ten-digit telephone number. The telephone company
also assigns a telephone number to each subscriber line connected
to each switch. The assigned telephone number includes the area
code and exchange code for the serving central office and four
unique digits.
Central office switches utilize office equipment (OE) numbers to
identify specific equipment such as physical links or circuit
connections. For example, a subscriber's line might terminate on a
pair of terminals on the main distribution frame of a switch 11.
The switch identifies the terminals, and therefore the particular
line, by an OE number assigned to that terminal pair. For a variety
of reasons, the operating company may assign different telephone
numbers to the one line at the same or different times. For
example, a local carrier may change the telephone number because a
subscriber sells a house and a new subscriber moves in and receives
a new number. However, the OE number for the terminals and thus the
line itself remains the same.
On a normal call, an end office type switch will detect an off-hook
condition on the line and provide dial tone. The switch identifies
the line by its OE number. The office also retrieves profile
information corresponding to the OE number and off-hook line. If
needed, the profile identifies the currently assigned telephone
number. The switch in the end office receives dialed digits and
routes the call. The switch may route the call to another line
serviced by that switch, or the switch may route the call over
trunks and possibly through one or more tandem offices to an office
that serves the called party's station or line. The switch
terminating a call to a destination will also utilize profile
information relating to the destination, for example to forward the
call if appropriate, to apply distinctive ringing, etc.
AIN call processing involves a query and response procedure between
an SSP capable switching office 11 and a database system, such as
the SCP 19. The SSP capable switching offices initiate such
processing upon detection of triggering events. At some point
during processing of a telephone call, a central office switching
system 11 will recognize an event in call processing as a `Point in
Call` (PIC) which triggers a query to the SCP 19. Ultimately, the
SCP 19 will return an instruction to the switching system 11 to
continue call processing. This type of AIN call processing can
utilize a variety of different types of triggers to cause the SSPs
11 to initiate the query and response signaling procedures with the
SCP 19. In the presently preferred embodiments discussed below, the
personal dial tone service utilizes an off-hook immediate trigger,
a dialed number trigger and a terminating attempt trigger (TAT), to
facilitate different aspects of the service.
In accord with one aspect of the present invention, before
providing dial-tone service, the SSP central office 11 that is
serving an outgoing call extends the call to the IP 23 providing
the speaker identification/verification (SIV) functionality. In the
preferred embodiments, this operation involves AIN type call
routing to the IP. The IP 23 prompts the caller and collects
identifying information, preferably in the form of speech. The IP
analyzes the caller's input to identify the caller as a particular
subscriber. If successful, the IP signals the SSP to load profile
data for that subscriber into the register assigned to the call in
the call store. In most of the preferred service applications, the
IP disconnects, and the SSP central office 11 processes the call in
accord with the loaded profile information. For example, the
central office 11 may now provide actual dial tone or provide a
message prompting the caller to dial a destination number. The
caller dials digits, and the central office processes the digits to
provide the desired outgoing call service, in the normal manner.
The IP may stay on the line, to monitor speech and thus caller
identity, for some service applications.
The call processing by the central office switch 11 utilizes the
loaded subscriber profile information. For example, the profile
data may indicate specific procedures for billing the call to this
subscriber on some account not specifically linked to the
originating telephone station or line. For example, in a college
dormitory, the billing information might specify billing of a
student's calls to the account of the student's parent(s). Any call
restrictions, imposed at the wish of the parents, would be
reflected in the profile. The switch would restrict the calling
services accordingly, e.g. to limit distance, cumulative cost
and/or duration of calls. The dormitory example is to be regarded
as merely illustrative of the varied situations to which the system
and methodology of the invention is applicable, as will become
apparent from following detailed description.
The inventors also envision use of selected subscriber profile
information on incoming calls. When a serving central office SSP 11
detects a call to a line having the personalized service,
processing hits a terminating attempt trigger (TAT). The SSP
interacts with the SCP 19 and routes the call to the IP 23. The IP
23 prompts the caller to identify a desired called party, e.g. one
of the students sharing the dormitory line. Menu announcement
together with either digit collection or preferably speech
recognition processing by the IP 23 facilitates identification of
the desired called party from those associated with the line. Based
on identification of the called subscriber, the IP 23 signals the
SSP switch 11 to load profile data for that subscriber into the
register assigned to the call in the call store. In this case,
however, the switch 11 uses selectively loaded profile information
for terminating the call. The IP disconnects, and the SSP central
office 11 processes the call in accord with the loaded profile
information.
For example, the central office 11 may provide a distinctive
ringing signal corresponding to the identified subscriber. This
service enables distinctive ringing for multiple subscribers on one
line without assigning each subscriber a separate telephone number.
The loaded profile information may specify call forwarding in event
of a busy or no-answer condition. This enables routing of the call
to the identified subscriber's mailbox, or another alternate
destination selected by the subscriber, even though the call did
not utilize a unique telephone number uniquely assigned to the
called subscriber.
It is a feature of one preferred embodiment of the invention that
the menu utilized on an incoming call also includes a so-called
`challenge` wherein the caller is requested to speak his or her
name. The profile of the called user which has been installed in
response to identification of the user may contain limitations
applicable to identified callers. To this end the speech
recognition node, preferably the IP, is provided with a previously
obtained template to permit identification of such callers. As
later described in further detail, the identification of both the
called and calling party may entail maintaining a voice connection
to the IP. Such a connection may be utilized for either recording
the conversation and/or bridging a third party onto the call, such
as a parent or other supervisory authority.
The present invention also encompasses a procedure in which a
subscriber calls in from a line not specifically designated for
personal dial tone service. The network routes the call to the IP
23, and the IP identifies the subscriber and the line from which
the subscriber called-in. The subscriber can interact with the IP
23 to have her personal dial tone service associated with that
line, either for one call or for some selected period of time. The
IP 23 instructs the appropriate central office switch(es) 11 to
load profile data associated with the subscriber.
The IP 23 might instruct the end office switch to load the profile
data only in the assigned call store register. The switch would use
the profile data only for a single call, for example to bill a call
from a pay-phone or a hotel room telephone to the subscriber's home
account. Alternatively, the IP 23 might instruct the central office
11 serving the line to the calling station 1 to utilize a virtual
office equipment number (OE) and associated profile data for calls
to and from that line for some period of time. In this later
example, the IP 23 would also instruct the central office 11
serving the line to the subscriber's home station 1 to modify the
subscriber's profile to forward calls for the subscriber's
telephone number. The modified profile data in the home office 11
would result in forwarding of the subscriber's incoming calls
through the office 11 to the selected station 1, for the set period
of time.
The present invention relies on the programmable functionality of
the central office switches and the enhanced call processing
functionalities offered by the IPs. To understand these various
functionalities, it may be helpful to review the structure and
operation of a program controlled central office and one
implementation of an IP. Subsequent description will explain
several of the above outlined call processing examples in greater
detail.
FIG. 2 is a simplified block diagram of an electronic program
controlled switch which may be used as any one of the SSP type
central offices 11 in the system of FIG. 1. As illustrated, the
switch includes a number of different types of modules. In
particular, the illustrated switch includes interface modules 51
(only two of which are shown), a communications module 53 and an
administrative module 55.
The interface modules 51 each include a number of interface units 0
to n. The interface units terminate lines from subscribers'
stations, trunks, T1 carrier facilities, etc. Each such termination
is identified by an OE number. Where the interfaced circuit is
analog, for example a subscriber loop, the interface unit will
provide analog to digital conversion and digital to analog
conversion. Alternatively, the lines or trunks may use digital
protocols such as T1 or ISDN. Each interface module 51 also
includes a digital service unit (not shown) which is used to
generate call progress tones and receive and detect dialed digits
in pulse code or dual-tone multifrequency form.
In the illustrated embodiment, the unit 0 of the interface module
51' provides an interface for the signaling and communication links
to the IP 23. In this implementation, the links preferably consist
of one or more ISDN PRI circuits each of which carries 23 bearer
(B) channels for communication traffic and one data (D) channel for
signaling data.
Each interface module 51 includes, in addition to the noted
interface units, a duplex microprocessor based module controller
and a duplex time slot interchange, referred to as a TSI in the
drawing. Digital words representative of voice information are
transferred in two directions between interface units via the time
slot interchange (intramodule call connections) or transmitted in
two directions through the network control and timing links to the
time multiplexed switch 57 and thence to another interface module
(intermodule call connection).
The communication module 53 includes the time multiplexed switch 57
and a message switch 59. The time multiplexed switch 57 provides
time division transfer of digital voice data packets between voice
channels of the interface modules 51 and transfers signaling data
messages between the interface modules. The switch 57 together with
the TSIs of the interface modules form the overall switch fabric
for selectively connecting the interface units in call
connections.
The message switch 59 interfaces the administrative module 55 to
the time multiplexed switch 57, so as to provide a route through
the time multiplexed switch permitting two-way transfer of control
related messages between the interface modules 51 and the
administrative module 55. In addition, the message switch 59
terminates special data links, for example a link for receiving a
synchronization carrier used to maintain digital synchronism.
The administrative module 55 provides high level control of all
call processing operations of the switch 11. The administrative
module 55 includes an administrative module processor 61, which is
a computer equipped with disc storage 63, for overall control of CO
operations. The administrative module processor 61 communicates
with the interface modules 51 through the communication module 53.
The administrative module 55 may include one or more input/output
processors (not shown) providing interfaces to terminal devices for
technicians and data links to operations systems for traffic,
billing, maintenance data, etc.
A CCIS terminal 73 and an associated data unit 71 provide an SS7
signalling link between the administrative module processor 61 and
one of the STPs 15 (see FIG. 1). Although only one such link is
shown, preferably there are a plurality of such links providing
redundant connections to both STPs of a mated pair and providing
sufficient capacity to carry all necessary signaling to and from
the particular office 11. The SS7 signaling through the terminal
73, the data unit 71 and the STPs provides two-way signaling data
transport for call set-up related messages to and from other
offices. These call set-up related messages typically utilize the
ISDN-UP (ISDN-users part) protocol portion of SS7. The SS7
signaling through the terminal 73, the data unit 71 and the STPs
also provides two-way signaling data transport for communications
between the office 11 and database systems or the like, such as the
SCP 19. The communications between the office 11 and the database
systems or the like utilize the TCAP (transactions capabilities
applications part) protocol portion of SS7.
As illustrated in FIG. 2, the administrative module 55 also
includes a call store 67 and a program store 69. Although shown as
separate elements for convenience, these are typically implemented
as memory elements within the computer serving as the
administrative module processor 61. The program store 69 stores
program instructions which direct operations of the computer
serving as the administrative module processor 61.
For each call in progress, a register assigned within the call
store 67 stores translation and user profile information retrieved
from disc storage 63 together with routing information and any
temporary information needed for processing the call. For example,
for a residential customer initiating a call, the call store 67
would receive and store line identification and outgoing call
billing information corresponding to an off-hook line initiating a
call. For the personal dial-tone service, the assigned register in
the call store 67 will receive and store different profile data
depending on the particular subscriber associated with any given
call. A register in the call store is assigned and receives profile
data from the disc memory both for originating subscribers on
outgoing calls and for terminating subscribers on incoming
calls.
A variety of adjunct processor systems known in the telephone
industry can be used as the IP 23. The critical requirements are
that the IP system process multiple calls and perform the
subscriber identification functions, preferably by speaker
identification and authentication. FIG. 3 is a functional diagram
illustration of an IP 23 for performing the subscriber
identification functions, possibly by dialed digit input and
preferably by analysis and recognition of speech.
The preferred IP architecture utilizes separate modules for
different types of services or functions, for example, one or two
Direct Talk type voice server modules 231A, 231B for interfacing
ISDN PRI trunks to the SSP central office(s) 11. Separate modules
233, 235 perform voice authentication and speech recognition. The
IP 23 includes a variety of additional modules for specific types
of services, such as a server module 237 for fax mail, and another
server 239 for voice mail services. The various modules communicate
with one another via an internal data communication system or bus
240, which may be an Ethernet type local area network.
Each Direct Talk module 231A or 231B comprises a general purpose
computer, such as an IBM RS-6000, having digital voice processing
cards for sending and receiving speech and other audio frequency
signals, such as IBM D-talk 600 cards. Each voice processing card
connects to a voice server card which provides the actual interface
to T1 or primary rate interface ISDN trunks to the switching
office. In the PRI implementation, the Direct Talk computer also
includes a signaling card, providing two-way signaling
communication over the D-channel of the PRI link. Each Direct Talk
computer also includes an interface card for providing two-way
communications over the internal data communications system
240.
The voice processing cards in the Direct Talk modules 231A, 231B
provide voice message transmission and dialed digit collection
capabilities. The modules 231A, 231B also perform the necessary
line interface functions for communications to and from those
servers which do not incorporate actual line interfaces. For
example, for facsimile mail, a Direct Talk module 231 connected to
a call would demodulate incoming data and convert the data to a
digital format compatible with the internal data communication
network 240. The data would then be transferred over network 240 to
the fax server 237. For outgoing facsimile transmission, the server
237 would transfer the data to one of the Direct Talk modules over
the network 240. The Direct Talk module 231 would reformat and/or
modulate the data as appropriate for transmission over the ISDN
link to the switch 11.
The Direct Talk modules provide a similar interface function for
the other servers, such as the voice mail server 239, the speech
recognition module 235 and the voice authentication module 233. For
incoming speech signals, the Direct Talk module connected to a call
receives digital speech signals in the standard pulse code
modulation format carried on a B-channel of an ISDN link. The
Direct Talk module reformats the speech data and transmits that
data over the internal network 240 to the server or module
performing the appropriate function, for example to the
authentication module 233 for analysis and comparison of features
to stored templates or feature data for known subscribers.
In the outgoing direction, the currently connected Direct Talk
module may play an announcement from memory, e.g. to prompt a
caller to say their name. Alternatively, the Direct Talk module may
receive digitized speech over the network 240 from one of the other
modules, such as a stored message retrieved from voice mail server
239. The Direct Talk module reformats the speech signal as needed
for transmission over the ISDN B-channel to the caller.
The illustrated IP also includes a communication server 243. The
communication server 243 connects between the data communication
system 240 and a router 241, which provides communications access
to the TCP/IP network 27 that serves as the second signaling
communication system. The communication server 243 controls
communications between the modules within the IP 23 and the second
signaling communication system. The server 243 and the router 241
facilitate communication between the elements of the IP 23 and the
SCP 19. The IP may also use this communication system to
communicate with other IP's, for example to send subscriber voice
template information to the remote IP 23.sub.R (FIG. 1) or to
receive such information from that IP or some other network
node.
The personalized service relies on the voice authentication module
233 to perform the necessary speaker identification/verification
function. For the identification and authentication of subscribers
or users, the voice authentication module 233 within the IP 23
stores a template or other feature or voice pattern information for
each person who has the personalized service in the area that the
IP services. For example, if the subscriber utilizes the personal
dial tone service from a particular line, such as a shared line in
a dormitory or the like, the IP stores the subscriber's voice
pattern information in a file associated with the office equipment
(OE) number of the particular line. If the IP 23 serving a call
does not store the template or feature data for a particular
subscriber, the IP 23 may obtain subscriber identification by
dialed digit input and then obtain a copy of the template or
feature data from a remote IP 23.sub.R via communication through
the TCP/IP network 27, in order to authenticate the subscriber's
identity.
Using current technology, a new subscriber or user would get on
line with the IP serving that subscriber and `train` that IP by
speaking certain phrases. From the received audio signals
representing those phrases, the IP would store templates or other
pattern information for use in identifying and/or verifying that a
caller is the particular subscriber.
During actual call processing, the voice authentication module 233
receives speech information from the caller. The voice
authentication module 233 compares the received information to its
stored template or feature data to identify a calling party as a
particular subscriber.
In the case of speech recognition applied to incoming calls, the IP
is trained in a different manner. Current speech recognition
technology permits recognition with a reasonable degree of
certitude based on training from a limited sample of recorded
speech of a subject. In situations where the target of the speech
recognition is not such as may participate in the cooperative
manner of subscribers, recorded samples of prior telephone speech
may be used with available recognition facilities of a more
sophisticated nature.
In such situations the present invention also relies on the speech
recognition capability of the module 235, particularly in
processing of incoming calls in certain situations. The speech
recognition module 235 enables the IP to analyze incoming audio
information to recognize vocabulary words. The IP 23 interprets the
spoken words and phrases to determine subsequent action. For
example, the IP might recognize the caller speaking the name of a
called subscriber and use the subscriber identification to instruct
the terminating central office to control the call in accord with
that subscriber's profile.
The preferred routing of the calls in accord with the invention
utilizes AIN type call processing. To understand the call
processing, it may be helpful to consider several specific examples
in more detail.
In a first example, consider an outgoing call from the station
1.sub.A to the station 1.sub.B. Assume per call assignment of
profile data to the originating line, for personal dial tone
service on each outgoing call. FIG. 4 provides a simplified flow
diagram of the signal flow and processing for such an outgoing
call.
Assume use of a standard telephone for purposes of this example.
The person lifts the handset creating an off-hook state in the
telephone 1.sub.A, and a corresponding signal or change in state on
the line to the central office 11 (step S1). In this call flow, the
off-hook signal is a type of service request, i.e. a request to
make an outgoing call. The serving central office 11.sub.1 detects
the off-hook and commences its call processing. Specifically, the
central office assigns a register in the call store 67 to this call
and loads profile information associated with the off-hook line
from the disc storage 63 into the assigned register. In this case,
the central office 11.sub.1 is an SSP capable office, and the
loaded profile data indicates an off-hook immediate trigger set
against the particular line. The serving SSP type office 11.sub.1
therefore detects this off-hook PIC as an AIN trigger (step
S2).
In response to the off-hook and the off-hook trigger set in the
subscriber's profile, the SSP type central office switch 11.sub.1
launches a query to the SCP 19 (step S3). Specifically, the SSP
11.sub.1 creates a TCAP query message containing relevant
information, such as the office equipment (OE) number assigned to
the off-hook line, and transmits that query over an SS7 link to one
of the STPs 15. The query includes a destination point code and/or
a global title translation addressing the message to the SCP 19,
and the STP 15 relays the query message over the appropriate link
to the SCP 19. The query from the SSP central office 11.sub.1
identifies the caller's line by its associated office equipment
(OE) number and possibly by a single telephone number associated
with the off-hook line.
In response to a query, the SCP 19 accesses its a database,
typically, the MSAP database set up in the ISCP, to determine how
to process the particular call. The SCP 19 identifies an access key
in the query and uses the key to retrieve the appropriate record
from the database. In this case, the query indicates an off-hook
trigger as the trigger event, therefore the SCP 19 uses the calling
party office equipment (OE) number as the access key. The SCP 19
retrieves a call processing record (CPR) corresponding to the
office equipment (OE) number associated with the off-hook line and
proceeds in accord with that CPR (step S4).
For the present example of the personal dial tone service, the CPR
will provide information necessary for routing the call to some
node of the network that will perform speaker
identification/verification (SIV). In the preferred embodiment, the
SIV is a function performed by an Intelligent Peripheral (IP),
therefore the CPR provides information for routing the call to the
nearest available IP having the SIV capability.
Based on the CPR, the SCP 19 formulates a response message
instructing the SSP central office 11.sub.1 serving the customer to
route the call. In this case, the message includes information,
e.g. a office equipment (OE) number or telephone number, used for
routing a call to the identified IP 23. The SCP 19 formulates a
TCAP message in SS7 format, with the destination point code
identifying the SSP office 11.sub.1. The SCP 19 transmits the TCAP
response message back over the SS7 link to the STP 15, and the STP
15 in turn routes the TCAP message to the SSP central office
11.sub.1 (see step S5) The SSP type switch in the central office
11.sub.1 uses the routing information to connect the call to one of
the lines or channels to the IP 23. A two-way voice grade call
connection now extends between the calling station 1.sub.A and the
IP 23 (step S6). In the present example, the switch actually
connects the off-hook line to the line to the IP before providing
dial tone.
As noted above, the communication link to the IP 23 provides both
line connections and signaling, preferably over a primary rate
interface (PRI) type ISDN link. When the central office 11.sub.1
extends the call from the calling party's line to a line circuit
(over a B channel) to the IP 23, the switch in that office also
provides call related data over the signaling link (D channel for
ISDN). The call related data, for example, includes the office
equipment (OE) number normally associated with the off-hook line
and possibly the telephone number for that line.
In response to the incoming call, the IP 23 will seize the line,
and it will launch its own query to the SCP 19 (step S7). In the
preferred network illustrated in FIG. 1, the IP 23 and the SCP 19
communicate with each other via a separate second signalling
network 27, for example utilizing either an 1129+ protocol or a
generic data interface (GDI) protocol as discussed in U.S. Pat. No.
5,572,583 to Wheeler, Jr. et al. The query from the IP 23 again
identifies the caller's line by at least its associated office
equipment (OE) number.
In response to the query from the IP 23, the SCP 19 again accesses
the appropriate CPR (step S8) and provides a responsive instruction
back through the network 27 to the IP 23 (step S9). Although the IP
23 could passively monitor any speech that the user might utter,
the preferred implementation utilizes a `Challenge Phase` to prompt
the user to input specific identifying information. In this case,
the instruction causes the IP 23 to provide a prompt message over
the connection to the caller (step S10). Here, the signal to the
caller may be a standard dial tone or any other appropriate audio
signal. Preferably, the instruction from the SCP 19 causes the IP
23 to provide an audio announcement prompting the caller to speak
personal information. In one preferred example, in step S10 the IP
plays an audio prompt message asking the caller, `Please say your
full name`. The process may ask for any appropriate identifying
information.
The signal received by the IP 23 goes over the lines and through
the central office switch(es) for presentation via the off-hook
telephone 1.sub.A to the calling party. In response, the caller
will speak identifying information into their off-hook telephone,
and the network will transport the audio signal to the IP 23 (step
S11).
As noted above, an IP 23 can provide a wide range of call
processing functions, such as message playback and digit
collection. In the preferred system, the IP also performs speaker
identification/verification (SIV) on the audio signal received from
the off-hook telephone in step S11. When the IP 23 receives speech
input information during actual call processing, for this service
example, the IP analyzes the speech to extract certain
characteristic information (step S12).
The IP 23 stores a template or other voice pattern information for
each person who has the personalized service in the area that the
IP normally services. If the IP 23 does not store the particular
template or feature information it needs to process a call, the IP
23 can communicate with a remote IP 23R to obtain that information.
In the present shared line example, the IP 23 will store template
or feature data for each subscriber associated with the particular
off-hook line.
When the IP 23 receives input speech and extracts the
characteristic information during actual call processing, the IP
compares the extracted speech information to stored pattern
information, to identity and authenticate the particular caller. In
the present example, the voice authentication module 233 in the IP
23 compares the extracted speech information to the stored template
or feature data for each subscriber associated with the particular
off-hook line.
In step S13, the IP 23 determines if the information extracted from
the speech input matches any of the stored template data feature
data for an identifiable subscriber (within some threshold level of
certainty). If there is a match, the IP now knows the identity of
the calling subscriber. Based on the identification of the calling
subscriber, the IP 23 selects a virtual office equipment (OE)
number from storage that corresponds to the subscriber.
The IP 23 formulates a D-channel signaling message containing the
virtual office equipment (OE) number together with an instruction
to load that OE number into the register assigned to the call in
place of the OE number of the off-hook line. The IP 23 supplies the
message to the SSP central office switch 11.sub.1 over the
D-channel of the ISDN PRI link (step S14). In response, the
administrative module processor 61 rewrites the OE number in the
register assigned to the call using the OE number received from the
IP 23.
Upon rewriting the OE number in the register, the administrative
module processor 61 of central office switch 11.sub.1 also reloads
the profile information in the register (step S15). Specifically,
the administrative module processor 61 retrieves profile
information associated with the virtual office equipment (OE)
number from the disc storage 63 into the register. As such, the
profile information in the assigned register in the call store 67
now corresponds to the identified subscriber, rather than to the
off-hook line.
The profile information provides a wide range of data relating to
the subscriber's services. The profile data provides necessary
billing information, enabling billing from the call to this
particular subscriber. The profile also defines various service
features available to this subscriber on outgoing calls, such as
three-way calling. The profile may define a class of calling
service available to the subscriber. In the dormitory example, the
caller may be allowed a set dollar amount for long distance calls
per month (e.g. $50.00). The profile data will indicate the
remaining amount at the time of the call and will cause the switch
to interrupt service when the available amount is exhausted. Other
class of service restrictions might enable long distance calls only
if collect and/or only if calling one or two specified numbers
(e.g. only to the parents' house). The class of service might
enable only long distance calls within a region or country but not
international calls.
In the presently preferred implementation, when the central office
switch 11.sub.1 reloads the profile, the central office disconnects
the link to the IP 23 and connects tone receivers to the caller's
line. Optionally, the central office 11.sub.1 may provide a `dial
tone` or other message over the line (step S16) The caller now
dials digits in the normal manner (step S17), and the switch in the
central office 11.sub.1 loads the dialed digits into the assigned
register within the call store 67. The central office 11.sub.1
utilizes the dialed digits and the subscriber's profile data to
process the call (S18). For example, if the dialed digits represent
a call within the subscriber's permitted class of service, the
switch completes the call to the destination station 1.sub.B using
the dialed digits in the normal manner. If the profile data
requires a particular billing treatment, e.g. to bill a long
distance call to the subscriber, the switch makes the appropriate
record and forwards the record to the exchange carrier company's
accounting office equipment. In accord with another aspect of the
invention, the network provides caller ID data naming the
identified subscriber to the destination station.
The processing to complete the call, performed in step S18,
actually involves a sequence of steps. Of particular note, some of
these steps facilitate delivery of caller ID information to the
destination station. The present invention involves delivering
caller ID information which corresponds to the identified
subscriber, preferably the subscriber's name, rather than simply
the number of the line or station from which the subscriber
initiates the call. Two processing methodologies are envisioned for
providing this calling subscriber ID feature, one involving access
to name information in a central database such as LIDB and the
other relying on name data from the subscriber's profile.
FIG. 4B is a simplified process and signal flow diagram,
illustrating the call completion operations, including caller ID
display using data from the profile. The network performs the steps
depicted in FIG. 4B after identification of the subscriber,
preferably based on speaker identification/verification (SIV). As
discussed earlier, the IP 23 supplies the signaling message
containing the virtual office equipment (OE) number and the
instruction to load that OE number into the assigned register to
the SSP central office switch 11.sub.1 over the D-channel of the
ISDN PRI link (step S14). In response, the administrative module
processor 61 rewrites the OE number in the register and reloads the
profile information in the register (step S15).
The central office 11.sub.1 provides dial tone or the like over the
line (step S16), the caller dials digits corresponding to the
desired destination (step S17), and the switch in the central
office 11.sub.1 begins is processing to route the call through the
network. Initially, the central office 11.sub.1 uses the dialed
number to initiate a CCIS communication with the exchange serving
the intended destination, in the example the terminating central
office 11.sub.N.
Specifically, the subscriber's serving central office 11.sub.1
generates an Initial Address Message (IAM) for transmission to the
terminating central office 11.sub.1 (S181). The IAM message
includes the SS7 destination point code (DPC) of the terminating
central office 11.sub.N and the SS7 origination point code (OPC) of
the customer's serving-end central office 11.sub.1, for addressing
purposes. The payload portion of the IAM message includes the
called and calling numbers. In accord with the invention, the
originating end office 11.sub.1 reads name data from the identified
subscriber's profile, currently loaded in the assigned register,
and places that data in additional field of the IAM message or in
an accompanying information message addressed in the same manner as
the IAM message. The originating central office transmits the IAM
message and possibly an accompanying information message through
the CCIS network to the distant terminating office 11.sub.N (step
S181).
When the terminating office 11.sub.N receives the IAM message, the
administrative module processor for that office retrieves the
customer profile for the number in the destination number field of
that message (e.g. the number for the telephone 1.sub.B) from its
mass storage system and loads that profile into one of its call
store registers. If the called party has an enhanced caller ID
service, with name display, the terminating central office 11.sub.N
would normally recognize the attempt to complete to that party's
number message as a terminating attempt trigger (TAT) type point in
call (PIC) to trigger access to the LIDB database for name
information. However, in this embodiment of the invention, the
terminating end office detects the receipt of the subscriber's name
data with the IAM message, therefore the administrative module
processor in that office overrides the trigger.
The terminating central office switching system 11.sub.N transmits
an Address Complete Message (ACM) back to the central office
11.sub.1 and if the called line is available applies ringing signal
to the called party's line (S182). The ACM includes a variety of
information, including a calling party status indicator, e.g. line
free or busy. If the line is not busy, the end office 13 rings the
station Y corresponding to the dialed digits 703-333-5678, and
generates the appropriate indicator in the Address Complete Message
(ACM) to indicate that it received the request for a call and that
the number is not busy. The ACM message is sent back by simply
reversing the point codes from the IAM message. Now the destination
point code (DPC) is the point code of the central office 11, and
the origination point code (OPC) is the point code of the central
office 13. In response to the ACM message, if the called line is
available, the originating central office 11 applies a ringback
tone signal to the line to the calling station 1.sub.A (S183).
As part of its operations to ring the called telephone station, the
terminating central office 11.sub.N transmits a caller ID signal
over the line. If the called party has ISDN service or the like,
the switch sends a signaling message along with the ringing signal.
If the called party has analog telephone service, the switch
11.sub.N transmits a caller ID message (step S184) as frequency
shift keyed (FSK) data inserted in the silent interval between the
first ringing signal (step S182) and the second ringing signal
(S185) applied to the called party's line.
In accord with the invention, the caller ID message applied to the
called party's line includes the telephone number associated with
the calling station 1.sub.A and at least some additional data
specific to the identified subscriber. If the called party has
enhanced caller ID for displaying name data, the ISDN telephone or
the caller ID terminal 5.sub.B receives the number and the name
data received with the IAM message in step S181. The caller ID
terminal 5.sub.B or a display device in the ISDN telephone displays
the received number and name information, identifying the actual
calling party, for review before the called party chooses to answer
the call.
If the called party subscribes only to normal caller ID, the end
office switch 11.sub.N can transmit only a limited amount of
information. For this purpose, the switch will select and transmit
one or two characters from the subscriber identification data along
with the telephone number. For example, if four persons normally
call from the particular originating telephone station or line, the
data sent to the terminating central office 11.sub.N might include
a letter or number identifying each subscriber. The switch 11.sub.N
would transmit that letter or number with the telephone number in
the caller ID message for display.
If someone answers the telephone station 1.sub.B, the terminating
central office switching system 11.sub.N detects an off-hook
condition (S13) and sends an Answer Message (ANM) back to the
originating central office 11.sub.1 through one or more of the STPs
15. The ANM message indicates that the called telephone 1.sub.B was
picked up. Also, at that time the actual telephone traffic trunk
circuit is connected together between the central offices 11.sub.1
and 11.sub.N. The central offices 11 connect the lines to the
stations to the respective ends of the trunk circuit, to complete
the voice path. At this point, actual voice communication is
established between the calling station 1.sub.A and the called
station 1.sub.B. Communication continues until one or both parties
hang up, at which time, all of the switched connections are torn
down.
FIG. 4C is a simplified process and signal flow diagram,
illustrating the call completion operations including, caller ID
display involving access to name information in a central LIDB
database. The network performs the steps depicted in FIG. 4C after
identification of the subscriber, preferably based on speaker
identification/verification (SIV). As in the example of FIG. 4B,
the central office switch 11.sub.1 receives an instruction
containing the subscriber's virtual office equipment (OE) number
(step S14), loads the corresponding profile information in the
register (step S15) and sends dial tone or the like over the line
(step S16). The subscriber dials digits corresponding to the
desired destination (step S17), and the switch in the central
office 11.sub.1 transmits an IAM message through the interoffice
signaling network to the terminating central office 11.sub.N. The
information sent in or with the IAM message in step S191, however,
is different than in the earlier example.
In this embodiment, the originating end office 11.sub.1 reads a
short code identifier from the identified subscriber's profile,
currently loaded in the assigned register, and places that
identifier in additional field of the IAM message or in an
accompanying information message addressed in the same manner as
the IAM message. For example, if the network provides personal dial
tone service to four identified persons associated with the
originating telephone 11.sub.1, the short code might comprise a
number from zero to three or letters such as A, B, C and D,
identified by the state of two bits in the IAM or accompanying
information message.
As in the earlier example, the originating end office 11.sub.1
addresses and transmits the IAM message with the specific
subscriber identifier code through the SS7 signaling network for
receipt by the terminating office 11.sub.N. If the called party has
only normal caller ID service, then the terminating office 11.sub.N
would transmit a normal caller ID message to the destination, with
the identifier appended to the calling party telephone number as an
extra digit or character. If the called party often receives calls
from this subscriber, even the limited subscriber specific
identification provided by the code will enable the called party to
recognize that the current call is from the identified
subscriber.
FIG. 4B depicts the processing steps, beginning in step S192, for
processing a call to a called customer having the enhanced caller
ID service for name and number display. In such a case, when the
terminating office 11.sub.N, the administrative module processor in
that office loads the profile for the called subscriber's telephone
number into a register in the call store assigned to this call. Of
particular note, because the called customer has the enhanced name
and number type caller ID service, the customer profile record
establishes a terminating attempt trigger (TAT) against the that
customer's telephone number.
At this point, the terminating office 11.sub.N recognizes the
called party telephone number in the destination number field of
the IAM message as a terminating attempt trigger (TAT) type point
in call or PIC (step S192). In response to this PIC, the
terminating office 11.sub.N launches a second query message through
one or more of the STP(s) 15 to the LIDB database 21 (step S193).
The query message includes both the telephone number associated
with the calling station 1.sub.A or its telephone line as well as
the code identifying the specific subscriber making that call.
The LIDB database 21 uses the calling party telephone number and
the code identifying the specific subscriber, received in the
query, to retrieve that one subscriber's account file record from
the database (step S194). The query also indicates the cause of the
query, i.e. the TAT triggering event. From this information, the
LIDB database recognizes that the query is a request for name
information. The database 21 therefore reads up to 15 characters of
name data from the subscriber's account file. The LIDB database 21
compiles a TCAP call control message including the name data and
returns that call control message to the terminating central office
11.sub.N via the SS7 network.
The terminating central office switching system 11.sub.N receives
the call control message from the LIDB database 21.
To provide the caller ID service in this embodiment, the
terminating end office 11.sub.N combines the name data from the
call control message together with the calling party number as two
caller ID messages. The end office 11.sub.N then signals the
originating office 11.sub.1 and initiates ringing of the called
party's line, as discussed in more detail below.
Assuming for this discussion that the called line is available, the
terminating central office switching system 11.sub.N transmits an
Address Complete Message (ACM) indicating availability back to the
central office 11.sub.1 and applies ringing signal to the called
party's line (step S182). In response to the ACM message, if the
called line is available, the originating central office 11 applies
a ringback tone signal to the line to the calling station 1.sub.A
(S183).
As part of its operations to ring the called telephone station, the
terminating central office 11.sub.N transmits a caller ID signal
over the line. If the called party has ISDN service or the like,
the switch sends the caller ID signaling messages along with the
ringing signal. If the called party has analog telephone service,
the switch 11.sub.N transmits the caller ID messages sequentially
over the line (step S184) as frequency shift keyed (FSK) data
inserted in the silent interval between the first ringing signal
(step S182) and the second ringing signal (S185) applied to the
called party's line. As in the earlier example, the display
provides the telephone number associated with the calling station
1.sub.A as well as the name data for the specifically identified
calling subscriber.
In the shared line example, each person normally expected to use
the line to station 1.sub.A is a different subscriber to the
personal dial tone service. As the subscribers make outgoing calls,
they each receive their own individualized service over the line on
each separate call, in precisely the manner described above
relative to steps S1 to S18 and the personal caller ID as described
above relative to FIGS. 4B and 4C. For example, each subscriber may
receive a different level of calling privileges and/or class of
service based on their ability and/or desire to pay for telephone
services. Also, the called party receives caller ID information
including both the origination telephone number and the name or
other identifying information associated specifically with the
calling subscriber.
Returning to step S13 in FIG. 4A, the extracted information
characterizing the input speech signals may not match any of the
templates or feature data used by the IP 23. In this event, the
process flows to step S19. The IP will count the number of tries or
attempts to identify the subscriber and permit some maximum number
of failed attempts (N). Assume, for example, that the software
allows only two identification attempts on one call (N=2). On the
first failure, the number of tries is less than N, therefore
processing returns to step S10, and the IP 23 again transmits the
prompt for speech input. The caller again speaks the requested
input information (S11), and the authentication module 233 again
analyzes the input information (S12). If the second input
adequately matches a stored subscriber's information in step S13,
the processing flows through steps S14 to S18 to complete the call
as described above.
However, if the extracted speech information does not match a
stored subscriber template or feature data, processing again flows
to step S19. If the number of tries now corresponds to the limit N,
for example on the second failed attempt, the processing branches
to step S20. The IP 23 may now transmit a message indicating denial
of service, although this is optional. If provided, the message
states that only a limited class of service is available in view of
the problems in recognizing the caller as a known subscriber.
The IP 23 formulates a D-channel signaling message instructing the
central office switch 11.sub.1 to process the call in accord with
default conditions and transmits that instruction to the central
office switch (step S21). The instruction could include a default
OE number corresponding to a default profile, or the message could
instruct the switch to proceed using the OE and profile data for
the off-hook line itself. The IP 23 supplies the message to the SSP
central office switch 11.sub.1 over the D-channel of the ISDN PRI
link (step S21). The administrative module processor 61 resumes
call processing using the appropriate default OE and profile.
In the preferred embodiment, the switch provides a normal dial tone
(S22), collects dialed digits from the caller (S23) and processes
the call (S24). However, the default profile provides only some
limited class of service, for example only emergency 911 service or
911 service plus flat rate local calling. The default call
processing provides no additional information from the profile
corresponding to any particular subscriber, therefore the network
processes the call as a normal call for caller ID purposes. The
caller ID service will provide only the telephone number to callers
having normal caller ID, and the network will access LIDB database
21 to provide name information if any associated strictly with the
telephone number, essentially in the manner that the network
provides such services when there is no personalized dial tone
service involved.
In the above example, the network disconnected the IP 23 after
identifying the subscriber and providing the subscriber's virtual
OE number to the serving central office 11. For some applications
of the personal dial tone service, the central office 11 would
maintain a bridged connection of the IP 23 on the line, to enable
the IP to monitor the call. For example, in a prisoner telephone
service, each prisoner would have only limited telephone rights as
specified in each prisoner's profile data. To prevent one prisoner
from selling their telephone service rights to another prisoner,
the IP 23 would periodically or constantly monitor the outgoing
speech signals from the prison line. The voice authentication
module 233 would initially identify the prisoner subscriber as
discussed above, and would periodically recheck to authenticate the
identity of the party using the prison line. If the voice
authentication module detects some other party using the line or
did not detect the identified subscriber's speech for some
predefined time interval, the IP 23 would instruct the serving
central office switch 11 to disconnect the call. The IP 23 may send
messages to the switch or to other network elements to initiate
additional action, such as profile modification to further limit a
particular prisoner's telephone privileges and/or to notify prison
authorities of misuse of telephone privileges.
If the called party subscribes only to normal caller ID, the end
office switch 11.sub.N can transmit only a limited amount of
information. For this purpose, the switch will select and transmit
one or two characters from the subscriber identification data along
with the telephone number. For example, if four persons normally
call from the particular originating telephone station or line, the
data sent to the terminating central office 11.sub.N might include
a letter or number identifying each subscriber. The switch 11.sub.N
would transmit that letter or number with the telephone number in
the caller ID message for display.
If someone answers the telephone station 1.sub.B, the terminating
central office switching system 11.sub.N detects an off-hook
condition (S13) and sends an Answer Message (ANM) back to the
originating central office 11.sub.1 through one or more of the STPs
15. The ANM message indicates that the called telephone 1.sub.B was
picked up. Also, at that time the actual telephone traffic trunk
circuit is connected together between the central offices 11.sub.1
and 11.sub.N. The central offices 11 connect the lines to the
stations to the respective ends of the trunk circuit, to complete
the voice path. At this point, actual voice communication is
established between the calling station 1.sub.A and the called
station 1.sub.B. Communication continues until one or both parties
hang up, at which time, all of the switched connections are torn
down.
FIG. 4C is a simplified process and signal flow diagram,
illustrating the call completion operations including, caller ID
display involving access to name information in a central LIDB
database. The network performs the steps depicted in FIG. 4C after
identification of the subscriber, preferably based on speaker
identification/verification (SIV) As in the example of FIG. 4B, the
central office switch 11.sub.1 receives an instruction containing
the subscriber's virtual office equipment (OE) number (step S14),
loads the corresponding profile information in the register (step
S15) and sends dial tone or the like over the line (step S16). The
subscriber dials digits corresponding to the desired destination
(step S17), and the switch in the central office 11.sub.1 transmits
an IAM message through the interoffice signaling network to the
terminating central office 11.sub.N. The information sent in or
with the IAM message in step S191, however, is different than in
the earlier example.
In this embodiment, the originating end office 11.sub.1 reads a
short code identifier from the identified subscriber's profile,
currently loaded in the assigned register, and places that
identifier in additional field of the IAM message or in an
accompanying information message addressed in the same manner as
the IAM message. For example, if the network provides personal dial
tone service to four identified persons associated with the
originating telephone 11.sub.1, the short code might comprise a
number from zero to three or letters such as A, B, C and D,
identified by the state of two bits in the IAM or accompanying
information message.
As in the earlier example, the originating end office 11.sub.1
addresses and transmits the IAM message with the specific
subscriber identifier code through the SS7 signaling network for
receipt by the terminating office 11.sub.N. If the called party has
only normal caller ID service, then the terminating office 11.sub.N
would transmit a normal caller ID message to the destination, with
the identifier appended to the calling party telephone number as an
extra digit or character. If the called party often receives calls
from this subscriber, even the limited subscriber specific
identification provided by the code will enable the called party to
recognize that the current call is from the identified
subscriber.
FIG. 4B depicts the processing steps, beginning in step S192, for
processing a call to a called customer having the enhanced caller
ID service for name and number display. In such a case, when the
terminating office 11.sub.N, the administrative module processor in
that office loads the profile for the called subscriber's telephone
number into a register in the call store assigned to this call. Of
particular note, because the called customer has the enhanced name
and number type caller ID service, the customer profile record
establishes a terminating attempt trigger (TAT) against the that
customer's telephone number.
At this point, the terminating office 11.sub.N recognizes the
called party telephone number in the destination number field of
the IAM message as a terminating attempt trigger (TAT) type point
in call or PIC (step S192). In response to this PIC, the
terminating office 11.sub.N launches a second query message through
one or more of the STP(s) 15 to the LIDB database 21 (step S193).
The query message includes both the telephone number associated
with the calling station 1.sub.A or its telephone line as well as
the code identifying the specific subscriber making that call.
The LIDB database 21 uses the calling party telephone number and
the code identifying the specific subscriber, received in the
query, to retrieve that one subscriber's account file record from
the database (step S194). The query also indicates the cause of the
query, i.e. the TAT triggering event. From this information, the
LIDB database recognizes that the query is a request for name
information. The database 21 therefore reads up to 15 characters of
name data from the subscriber's account file. The LIDB database 21
compiles a TCAP call control message including the name data and
returns that call control message to the terminating central office
11.sub.N via the SS7 network.
The terminating central office switching system 11.sub.N receives
the call control message from the LIDB database 21.
To provide the caller ID service in this embodiment, the
terminating end office 11.sub.N combines the name data from the
call control message together with the calling party number as two
caller ID messages. The end office 11.sub.N then signals the
originating office 11.sub.1 and initiates ringing of the called
party's line, as discussed in more detail below.
Assuming for this discussion that the called line is available, the
terminating central office switching system 11.sub.N transmits an
Address Complete Message (ACM) indicating availability back to the
central office 11.sub.1 and applies ringing signal to the called
party's line (step S182). In response to the ACM message, if the
called line is available, the originating central office 11 applies
a ringback tone signal to the line to the calling station 1.sub.A
(S183).
As part of its operations to ring the called telephone station, the
terminating central office 11.sub.N transmits a caller ID signal
over the line. If the called party has ISDN service or the like,
the switch sends the caller ID signaling messages along with the
ringing signal. If the called party has analog telephone service,
the switch 11.sub.N transmits the caller ID messages sequentially
over the line (step S184) as frequency shift keyed (FSK) data
inserted in the silent interval between the first ringing signal
(step S182) and the second ringing signal (S185) applied to the
called party's line. As in the earlier example, the display
provides the telephone number associated with the calling station
1.sub.A as well as the name data for the specifically identified
calling subscriber.
In the shared line example, each person normally expected to use
the line to station 1.sub.A is a different subscriber to the
personal dial tone service. As the subscribers make outgoing calls,
they each receive their own individualized service over the line on
each separate call, in precisely the manner described above
relative to steps S1 to S18 and the personal caller ID as described
above relative to FIGS. 4B and 4C. For example, each subscriber may
receive a different level of calling privileges and/or class of
service based on their ability and/or desire to pay for telephone
services. Also, the called party receives caller ID information
including both the origination telephone number and the name or
other identifying information associated specifically with the
calling subscriber.
Returning to step S13 in FIG. 4A, the extracted information
characterizing the input speech signals may not match any of the
templates or feature data used by the IP 23. In this event, the
process flows to step S19. The IP will count the number of tries or
attempts to identify the subscriber and permit some maximum number
of failed attempts (N). Assume, for example, that the software
allows only two identification attempts on one call (N=2). On the
first failure, the number of tries is less than N, therefore
processing returns to step S10, and the IP 23 again transmits the
prompt for speech input. The caller again speaks the requested
input information (S11), and the authentication module 233 again
analyzes the input information (S12). If the second input
adequately matches a stored subscriber's information in step S13,
the processing flows through steps S14 to S18 to complete the call
as described above.
However, if the extracted speech information does not match a
stored subscriber template or feature data, processing again flows
to step S19. If the number of tries now corresponds to the limit N,
for example on the second failed attempt, the processing branches
to step S20. The IP 23 may now transmit a message indicating denial
of service, although this is optional. If provided, the message
states that only a limited class of service is available in view of
the problems in recognizing the caller as a known subscriber.
The IP 23 formulates a D-channel signaling message instructing the
central office switch 11.sub.1 to process the call in accord with
default conditions and transmits that instruction to the central
office switch (step S21). The instruction could include a default
OE number corresponding to a default profile, or the message could
instruct the switch to proceed using the OE and profile data for
the off-hook line itself. The IP 23 supplies the message to the SSP
central office switch 11.sub.1 over the D-channel of the ISDN PRI
link (step S21) The administrative module processor 61 resumes call
processing using the appropriate default OE and profile.
In the preferred embodiment, the switch provides a normal dial tone
(S22), collects dialed digits from the caller (S23) and processes
the call (S24). However, the default profile provides only some
limited class of service, for example only emergency 911 service or
911 service plus flat rate local calling. The default call
processing provides no additional information from the profile
corresponding to any particular subscriber, therefore the network
processes the call as a normal call for caller ID purposes. The
caller ID service will provide only the telephone number to callers
having normal caller ID, and the network will access LIDB database
21 to provide name information if any associated strictly with the
telephone number, essentially in the manner that the network
provides such services when there is no personalized dial tone
service involved.
In the above example, the network disconnected the IP 23 after
identifying the subscriber and providing the subscriber's virtual
OE number to the serving central office 11. For some applications
of the personal dial tone service, the central office 11 would
maintain a bridged connection of the IP 23 on the line, to enable
the IP to monitor the call. For example, in a prisoner telephone
service, each prisoner would have only limited telephone rights as
specified in each prisoner's profile data. To prevent one prisoner
from selling their telephone service rights to another prisoner,
the IP 23 would periodically or constantly monitor the outgoing
speech signals from the prison line. The voice authentication
module 233 would initially identify the prisoner subscriber as
discussed above, and would periodically recheck to authenticate the
identity of the party using the prison line. If the voice
authentication module detects some other party using the line or
did not detect the identified subscriber's speech for some
predefined time interval, the IP 23 would instruct the serving
central office switch 11 to disconnect the call. The IP 23 may send
messages to the switch or to other network elements to initiate
additional action, such as profile modification to further limit a
particular prisoner's telephone privileges and/or to notify prison
authorities of misuse of telephone privileges.
The first detailed example discussed above related to personal dial
tone service provided on a per-call basis on a shared use line.
Several known subscribers might routinely use their personal dial
tone service over the same line. As noted earlier, an alternate
form of the personal dial tone service can be activated on a
dial-up basis. Consider now an example of a dial-up activation for
a single call.
For this example, assume that a subscriber's normal or `home`
telephone is telephone 1.sub.B. The end office switch 11.sub.N
stores the subscriber profile data for the line associated with
that telephone station. Now assume that the subscriber is using
station 1.sub.A connected through a telephone line to central
office 11.sub.1. FIG. 5 provides a simplified flow diagram of the
signal flow and processing for such a call.
The subscriber lifts the handset creating an off-hook state in the
telephone 1.sub.A and a signal to office 11 (step S31). The serving
central office 11.sub.1 detects the off-hook and commences its call
processing. Specifically, the central office assigns a register in
the call store 67 to this call and loads profile information
associated with the off-hook line from the disc storage 63 into the
register. In this case, the profile data associated with the line
does not provide an off-hook trigger because the line is not
specifically associated with the shared use type personal dial tone
service discussed above. The central office 11.sub.1 therefore
provides dial tone in the normal manner (step S32).
If making a normal call, the caller would dial a destination
number, and the network would complete the call as dialed. To
activate the personal dial tone service, however, the subscriber
dials an access number assigned to that service, such as
1-800-DIALTON, from the station 1.sub.A (step S33).
The dialing of an outgoing call, in this case to the access number,
is another type of service request. The central office switch
11.sub.1 recognizes the dialed access number as a trigger event or
`PIC` (step S34). The SSP type central office 11.sub.1 creates a
TCAP query message containing relevant information, such as the
office equipment (OE) number and/or telephone number assigned to
the off-hook line, the dialed number and the type of triggering
event. The office 11.sub.1 transmits that query to the SCP 19 (step
S35). Specifically, the SSP central office 11.sub.1 transmits the
query over an SS7 link to one of the STPs 15. The query includes a
point code and/or a global title translation addressing the message
to the SCP 19, and the STP 15 relays the query message over the
appropriate link to the SCP 19.
In response to a query, the SCP 19 accesses its database to
determine how to process the particular call. In this case, the
query indicates the dialed number type trigger and provides the
digits of the specific number dialed. The SCP 19 uses the dialed
number as the access key. The SCP 19 retrieves a call processing
record (CPR) corresponding to that number associated with the
personal dial tone access function (step S36). For the current
exemplary access, the CPR will provide information necessary for
routing the call to the IP 23 that will perform the necessary
speaker identification/verification (SIV).
Based on the CPR, the SCP 19 formulates a response message
instructing the SSP central office 11.sub.1 serving the customer to
route the call. In this case, the message includes information,
e.g. a office equipment (OE) number or telephone number, used for
routing a call to the identified IP 23. The SCP 19 formulates a
TCAP response in SS7 format and transmits the TCAP response message
back to the SSP central office 11.sub.1 (see step S37).
The SSP type switch in the central office 11.sub.1 uses the routing
information to connect the call to a line or channel to the IP 23.
A voice grade call connection now extends between the calling
station 1.sub.A and the IP 23 (step S38).
The central office 11 provides a signaling message to the IP 23
with the call. In this case, the signaling message includes the
dialed digits indicating a call to the personal dial tone access
number. The signaling message also includes either the office
equipment number or the telephone number of the line to the calling
station 1.sub.A.
As in the earlier example, the IP 23 will seize the line for the
incoming call and launch a query to the SCP 19 through the TCP/IP
network 27 (step S39). The SCP 19 accesses an appropriate CPR
(S40), and based on that CPR, the SCP 19 transmits back a message
(S41) instructing the IP 23 to execute a program or script for the
dial-up access to the personal dial-tone service.
The IP initially plays a greeting and a prompt message (S42) and
collects spoken input information (S43). The IP 23 may also play a
prompt and collect digits representing the subscriber's normal or
home telephone number. The voice authentication module 233 analyzes
the spoken identification information to extract characteristic
information (S44) and compares the extracted information to stored
template or feature data to determine if there is an adequate match
to the known subscriber data (S45), as in the earlier example.
In step S45, the IP 23 determines if the information extracted from
the speech input matches any of the stored template data feature
data for an identifiable subscriber. If there is a match, the IP
now knows the identity of the calling subscriber. Based on the
identity of the subscriber, the IP 23 obtains the subscriber's
profile data from the central office 11.sub.N serving the
subscriber's home telephone line. If the IP 23 is in direct
signaling communication with the home central office 11.sub.N, for
example via an ISDN D-channel or an SMDI link, the IP 23 may
directly request and receive the profile data over the signaling
link. If the IP and the switch are not in direct communication, the
IP may provide a message notifying the SCP 19, and the SCP 19 would
obtain the data from the switch and provide it back to the IP
23.
The IP 23 formulates a D-channel signaling message containing the
subscriber's profile information together with an instruction to
load that information into the register assigned to the call in
place of the profile information corresponding to the off-hook line
(step S46). The IP 23 supplies the message to the SSP central
office switch 11.sub.1 over the D-channel of the ISDN PRI link. In
response, the administrative module processor 61 rewrites the
profile data in the register assigned to the call using the data
from the IP 23 (step S47). As such, the profile information in the
assigned register now corresponds to the identified subscriber.
When the central office switch 11.sub.1 reloads the profile, the
central office disconnects the link to the IP 23 and connects tone
receivers to the caller's line. The central office 11.sub.1 may
also provide a standard dial tone or other message over the line
(step S48). The caller can now dial digits in the normal manner
(step S49), and the switch in the central office 11.sub.1 will load
the dialed digits into the assigned register within the call store
67. The central office 11.sub.1 utilizes the dialed digits and the
subscriber's profile data to process the call (step S50). For
example, the switch in central office 11.sub.1 may provide the
appropriate record to bill the outgoing call to the subscriber's
account. In accord with the invention, the network also provides
the subscriber specific information for caller ID purposes, in the
manner discussed in detail above relative to either FIG. 4B or FIG.
4C.
As in the earlier example, the preferred embodiment allows up to N
tries or attempts to provide recognizable subscriber identification
information. Thus, if in step S45 the extracted information
characterizing the input speech signals did not match any of the
templates or feature data used by the IP 23, then the process flows
to step S51. If the current number of attempts for recognition on
this call is less than N, processing returns to step S42, and the
IP 23 again transmits the prompt for speech input. The caller again
speaks the requested input information (S43), and the
authentication module 233 again analyzes the input information
(S44). If the second input adequately matches a stored subscriber's
information S45, the processing flows through steps S46 to S50 to
complete the call as described above.
However, if the extracted speech information does not match a
stored subscriber template or feature data, processing again flows
to step S51. If the number of tries now corresponds to the limit N,
the processing branches to step S52. The IP 23 preferably transmits
a message indicating denial of service (S52), and then transmits a
message to the central office 11.sub.1 signifying disconnection of
the access call (S53). It should be noted that, in this example,
normal service provided over the line to station 1.sub.A is
available on a subsequent call. The failure to recognize the caller
as a personal dial tone subscriber only prevents the caller from
using the personal dial tone services of a subscriber to that
service, for example specialized billing of calls to that
subscriber's account instead of to the account normally associated
with the line to the calling station 1.sub.A.
In the above discussed dial-up access example, the dial tone
service was personalized for a single outgoing call by temporarily
loading the subscriber's profile data into the register assigned to
the outgoing call in the originating central office 11.sub.1. The
system can provide such service to the subscriber over any line or
to any telephone station, including pay telephone stations.
The present invention also enables activation of the personal dial
tone service on a particular line for some predetermined period of
time, for example to enable use of office or business services from
some remote location while a business subscriber is out of the
office. This type of operation involves an activation call
requesting the service on a particular line for the desired period.
Consider now an example of such a time activated service.
For this example, assume that a subscriber's normal business
telephone is telephone 1.sub.B. The end office switch 11.sub.N
stores the subscriber profile data for the line associated with
that telephone station. Now assume that the subscriber is using
station 1.sub.A connected through a telephone line to central
office 11.sub.1 for business related communication services. The
business related communication services include both incoming call
related services and outgoing call related services.
To activate the personal dial tone service, the subscriber again
lifts the handset at station 1.sub.A, receives dial tone from the
central office 11, and dials the access number assigned to that
service. The network uses AIN type processing to route the call to
the IP 23, as in the example discussed above relative to FIG.
5.
As in the earlier examples, the IP 23 seizes the line for the
incoming call and launches a query to the SCP 19 through the TCP/IP
network 27. The SCP 19 transmits back a message instructing the IP
23 to play a greeting and a prompt message and collect and analyze
spoken input information to identify and authenticate the
subscriber. The instruction from the SCP 19 also causes the IP 23
to prompt the subscriber and obtain input information regarding the
time period for service activation and possibly to obtain digits
representing the subscriber's normal business telephone number. The
process of calling the access number and interacting with the IP to
activate the personal dial tone service on a line for the desired
period is another type of service request.
For outgoing call processing, the IP 23 signals the central office
11.sub.1 serving the line to station 1.sub.A to set an off-hook
trigger in the profile data associated with that line. The IP also
obtains the profile information from the switch 11.sub.N serving
the station 1.sub.B and provides that information together with a
virtual OE number to the central office 11.sub.1. The office
11.sub.1 stores the profile in its disc memory 63 in such a manner
that the switch in that office can use the virtual OE number to
retrieve that subscriber's profile. For incoming calls to the
subscriber, the IP 23 transmits a signaling message to the
subscriber's home office 11.sub.N to set a terminating attempt
trigger (TAT) against the line to the subscriber's office telephone
1.sub.B.
The IP 23 also transmits a message through the TCP/IP network 27 to
the SCP 19 advising the SCP 19 of the service activation. This
message identifies the subscriber, for example by their normal
telephone number and identifies the telephone number and office
equipment (OE) number associated with the line to station 1.sub.A
that the subscriber selected for their personal dial tone
service.
In response to the message from the IP 23, the SCP 19 now
establishes or modifies two CPRs for this subscriber. One CPR
controls processing of calls to the subscriber's normal business
telephone number to enable routing to the station 1.sub.A, and the
other controls routing of outgoing calls from that station to the
IP 23 for speaker identification/verification (SIV) processing.
Subsequently, when there is an outgoing call from the station
1.sub.A, the network will route the call to the IP 23 to determine
if the caller is the subscriber or some other party, exactly as
discussed in the per-call service from a shared use line (FIG. 4).
As in that earlier example, if the IP identifies the caller as the
personal dial tone subscriber, then the IP 23 provides the virtual
OE number to enable loading of subscriber's profile from disc
memory 63. The network provides the telephone number and the
subscriber specific information, for caller ID purposes, as
discussed above. If the IP determines that the caller is not the
personal dial tone subscriber, the IP instructs the originating
office 11.sub.1 to simply provide dial tone and complete the call
in the normal manner. The central office 11.sub.1 therefore will
utilize the office equipment (OE) number and profile information
normally associated with the line, instead of those for the
personal dial tone subscriber. The network provides caller ID
service, identifying the number and possibly the main name
associated with the line, in the normal manner. In this way, it is
quite easy for the personal dial tone subscriber and the normal
subscriber to both obtain their desired services on their
respective calls via the same line, and to be correctly identified
to called parties who subscribe to caller ID services.
The trigger set against the subscriber's normal telephone number
and establishment of the CPR in the SCP 19 enables redirection of
calls normally intended for the subscriber's business telephone
1.sub.B to the line to station 1.sub.A. Depending on how the
subscriber elects to define their individual service, the network
may simply route the calls to the line to station 1.sub.A, as a
normal AIN forwarded call that simply rings the station(s) 1.sub.A
on the line. Alternatively, the subscriber may elect an enhanced
service which involves routing to the IP, IP prompting and speech
recognition to identify the called subscriber and distinctive
ringing over the line, in a manner analogous to that used for
processing incoming calls in shared use applications, such as the
above discussed dormitory example.
As noted above, the dial-up access procedure in this latest service
example required the subscriber to specify a time period that the
personal dial tone service should apply to the particular line. The
IP 23 stores a record of the time period elected by the subscriber.
When the period expires or if the subscriber calls in earlier to
change the service to another line or temporarily cancel the
service, the IP 23 will provide cancellation notices to the
appropriate central offices 11 and to the SCP 19. In the example,
the IP 23 will notify the office 11.sub.1 to cancel the off-hook
trigger set against the line to station 1.sub.A and to delete the
subscriber's virtual OE number and profile from its disc memory.
The IP 23 will also instruct the central office 11.sub.N to cancel
the terminating attempt trigger set against the subscriber's
business line to station 1.sub.B. The notice to the SCP 19 causes
the SCP to deactivate the personal dial tone CPR and the call
redirection CPR. If the associated personal identification
functionality for caller ID service relies on a central database,
such as LIDB, the IP would also instruct that database to
temporarily establish a subscriber account record associating the
subscriber's name and calling card billing information with the
telephone number and a subscriber identifier code.
The subscriber can then or later interact with the IP 23 to
establish time based temporary personal dial tone service through
another line or location, as discussed above. In this manner, a
subscriber might set up a temporary office in a motel in one city
for several days. The subscriber might cancel the service while in
transit to a new location. Then the subscriber might reestablish
the service to set up a temporary office service at a vacation home
for a week.
The time based personal dial tone service could be modified in
several manners. For example, the subscriber might establish a file
for use by the SCP or the IP to establish the personal dial tone
service at two or more locations at specified times, e.g. at the
office during office hours and at a home office during other hours.
Also, the above example of this service relied on downloading the
subscriber' profile into the switch serving the line with which the
subscriber is temporarily associated. Alternatively, the IP could
obtain the profile from the subscriber's home switch and provide
the profile to the serving switch as part of the processing of each
outgoing call by the subscriber from that line during the specified
time period.
A preferred network implementation and a number of specific call
processing routines have been discussed above by way of examples
relating to the present invention. However, the preferred
embodiment of the invention is amenable to a variety of
modifications.
For example, the preferred embodiment described above utilizes
speaker identification/verification to recognize the identity of a
calling subscriber. Where such capabilities are not available, the
system could use an announcement and digit collection process, for
example to obtain an account number and a personal identification
number (PIN).
Also, the currently preferred embodiment utilizes AIN routing to
the IP and speaker identification/verification elements within the
IP to identify the subscriber for profile selection. As speaker
identification/verification equipment becomes more readily
available, cheaper and more compact, it will be possible to build
this functionality into the line cards of the end office switches.
The switch itself will challenge the caller, analyze spoken
information and identify the subscriber to select the appropriate
profile, without routing to an IP or the like.
While the foregoing embodiments of the invention supply many
outstanding needs, there still exists a need for a method of
conveniently and economically coping with a number of problems
which manifest themselves in one or another objecionable type of
usage of the public telephone network. These may comprise usages
which are either illegal or detrimental to the health, safety and
security of Telco subscribers. By way of example, one problem of
widespread significance is the provision of adequate protection of
the security and well being of so called "latchkey children." As
will be understood, this term is applied to children, usually of
school age, who have working parents but who arrive home from
school prior to the return of their parents.
These children admit themselves to their residence or premises and
are usually instructed by their parents to keep the door locked or
latched until a parent returns. In addition to these instructions
parents usually admonish such children to follow parent prescribed
rules in answering or using the telephone. However, experience has
demonstrated that the telephone is still subject to usages which
pose threats of one or another types to the children. The problem
is most acute where multiple children are housed with a single
telephone link to the customer premises.
Parents or guardians usually provide each child with a list of
permitted calls. For example, any of the children may be permitted
to call 911 in case of emergency. All of the children may be
permitted to call designated relatives or friends of the family.
However, the call permissions and restrictions usually vary from
child to child. The older children may be allowed calls to
designated schoolmates or friends. The identity of the parties to
whom the children are permitted to place calls varies with the
identity of the child. Conversely each child may have individually
prohibited calls. In the usual situation all calls which are not
expressly designated as allowed will be prohibited.
In addition to this list of permitted outgoing calls, the children
are usually provided with specific instructions as to calls to be
answered. However this is difficult to regulate, even in the case
of obedient children. For example, the availability of a Caller ID
service offers no guarantee that the indicated caller is actually
on the line. The present preferred embodiment of the invention
provides a system and method for supplying this need.
Following is a description to the operation of one preferred
embodiment of the invention which addresses the problem of
providing implementation of the instructions of the parents or
guardians with a reasonable degree of certitude.
In this example it is assumed that there is a subscriber premise
which houses a pair of latchkey children A and B. Child A and child
B have each been provided with a list of one or more incoming calls
which they are permitted to receive. Each such child (hereinafter
sometimes referred to as a subscriber) is provided with a personal
customer profile record which is identified by a virtual OE number.
Each such profile contains data which specifies permissible and
prohibited communications for the individual child. For example,
each customer profile identifies the callers whose calls may be
accepted.
The central office switch identifies the particular line, by the OE
number assigned to that line and line number. The switch also
stores and retrieves profile data which it stores for that line and
number and that profile data reflects the special services to which
that line and number is subscribed. When the central office detects
a call to a line having the personalized service, processing hits a
terminating attempt trigger (TAT). The SSP switch interacts with
the SCP and routes the call to the IP. The IP prompts the caller to
identify a desired called party, e.g. one of the children sharing
the line. Menu announcement together with either digit collection
or preferably speech recognition processing by the IP facilitates
identification of the desired called child from others associated
with the line. Based on identification of the called child, the IP
signals the SSP switch to load profile data for that specific child
into the register assigned to the call in the call store.
This substitution is accomplished. In this case, the profile for
child A contains data information which indicates that child A is
permitted to accept a call from child C but that child C is
required to authenticate herself. The IP is apprised of this
requirement and uses another prompt to the calling party to
identify herself. This may be a prompt such as "Who is calling?". A
template for the voice of child C is maintained in the IP. This
template is now used by the IP to verify that the caller is in fact
child C. Child C has now been identified and authenticated as the
calling party.
The profile for child A may provide that a distinctive ringing
signal is to be used corresponding to the identified subscriber or
child A. In this event distinctive ringing for child A is used to
attempt to have child A answer the telephone. According to the
loaded profile for child A, the answering party is prompted to
speak her name. The IP remains bridged onto the connection and uses
voice processing to verify a match between the spoken response and
a template previously installed in the IP. Assuming verification,
the switch concludes processing of the call in accord with the
loaded profile information, i.e., makes the connection and permits
the voice communication to occur. The IP is disconnected.
If the initial authentication of the calling party fails, i.e., if
the caller states her name to be that of child C but the voice
verification fails to confirm a match, the calling party may be
permitted one or more additional attempts. If these fail, the
invention comprehends a plurality of consequential handling
steps.
In the simplest case the call attempt is terminated by
disconnection of the calling line or link. As another option, the
incoming call may be forwarded to a third party line, such as a
pre-designated line to a parent or guardian. In instances
satisfying applicable provisions of law, the parent or guardian may
record the ensuing dialogue. The specific handling which is
performed is contained in the data information in the profile which
has been personalized for the subscriber represented by child
A.
The foregoing example has dealt with affording protection to
latchkey children in the case of incoming calls. It is a further
feature of the invention that the invention provides a system for
preventing the initiation of proscribed outgoing calls from the
subscriber premise and line. Following is an example of the
operation of such protection.
As previously stated, the preferred routing of the calls in accord
with the invention utilizes AIN type call processing. In the case
of one of the children A or B initiating a call the phone goes
off-hook. The serving central office 11.sub.1 detects the off-hook
and commences its call processing. Specifically, the central office
assigns a register in the call store 67 to this call and loads
profile information associated with the off-hook line from the disc
storage 63 into the assigned register. In this case, the central
office 11.sub.1 is an SSP capable office, and the loaded profile
data indicates an off-hook immediate trigger set against the
particular line. The serving SSP type office 11.sub.1 therefore
detects this off-hook PIC as an AIN trigger.
In response to the off-hook and the off-hook trigger set in the
subscriber's profile, the SSP type central office switch 11.sub.1
launches a query to the SCP 19. Specifically, the SSP 11.sub.1
creates a TCAP query message containing relevant information, such
as the office equipment (OE) number assigned to the off-hook line,
and transmits that query over an SS7 link to one of the STPs
15.
The STP 15 relays the query message over the appropriate link to
the SCP 19. The query from the SSP central office 11.sub.1
identifies the caller's line by its associated office equipment
(OE) number and possibly by a single telephone number associated
with the off-hook line.
In response to a query, the SCP 19 accesses its a database,
typically, the MSAP database set up in the ISCP, to determine how
to process the particular call. The SCP 19 identifies an access key
in the query and uses the key to retrieve the appropriate record
from the database. In this case, the query indicates an off-hook
trigger as the trigger event, therefore the SCP 19 uses the calling
party office equipment (OE) number as the access key. The SCP 19
retrieves a call processing record (CPR) corresponding to the
office equipment (OE) number associated with the off-hook line and
proceeds in accord with that CPR.
The CPR will provide information necessary for routing the call to
some node of the network that will perform speaker
identification/verification (SIV), in this example the SIV is a
function performed by an Intelligent Peripheral (IP). Therefore the
CPR provides information for routing the call to the nearest
available IP having the SIV capability.
Based on the CPR, the SCP 19 formulates a response message
instructing the SSP central office 11.sub.1 serving the customer to
route the call. In this case, the message includes information,
e.g. a office equipment (OE) number or telephone number, used for
routing a call to the identified IP 23. The SCP 19 formulates a
TCAP message in SS7 format, with the destination point code
identifying the SSP office 11.sub.1. The SCP 19 transmits the TCAP
response message back over the SS7 link to the STP 15, and the STP
15 in turn routes the TCAP message to the SSP central office
11.sub.1.
The SSP type switch in the central office 11.sub.1 uses the routing
information to connect the call to one of the lines or channels to
the IP 23. A two-way voice grade call connection now extends
between the calling station 1.sub.A and the IP 23. In the present
example, the switch actually connects the off-hook line to the line
to the IP before providing dial tone.
As noted above, the communication link to the IP 23 provides both
line connections and signaling, preferably over a primary rate
interface (PRI) type ISDN link. When the central office 11.sub.1
extends the call from the calling party's line to a line circuit
(over a B channel) to the IP 23, the switch in that office also
provides call related data over the signaling link (D channel for
ISDN). The call related data, for example, includes the office
equipment (OE) number normally associated with the off-hook line
and possibly the telephone number for that line.
In response to the incoming call, the IP 23 will seize the line,
and it will launch its own query to the SCP 19 (step S7). In the
preferred network illustrated in FIG. 1, the IP 23 and the SCP 19
communicate with each other via a separate second signaling network
27, for example utilizing either an 1129+ protocol or a generic
data interface (GDI) protocol. The query from the IP 23 again
identifies the caller's line by at least its associated office
equipment (OE) number.
In response to the query from the IP 23, the SCP 19 again accesses
the appropriate CPR and provides a responsive instruction back
through the network 27 to the IP 23. Although the IP 23 could
passively monitor any speech that the user might utter, the
preferred implementation utilizes a `Challenge Phase` to prompt the
user to input specific identifying information. In this case, the
instruction causes the IP 23 to provide a prompt message over the
connection to the caller. Here, the signal to the caller is
preferably an audio announcement prompting the caller to speak
personal information. In one preferred example, the IP plays an
audio prompt message asking the caller, `Please say your name`. The
process may ask for any appropriate identifying information.
The signal received by the IP 23 goes over the lines and through
the central office switch(es) for presentation via the off-hook
telephone 1.sub.A to the calling party. In response, the caller
will speak identifying information into their off-hook telephone,
and the network will transport the audio signal to the IP 23. When
the IP 23 receives speech input information during actual call
processing, for this service example, the IP analyzes the speech to
extract certain characteristic information.
As previously explained, the IP 23 stores a template or other voice
pattern information for each person who has the personalized
service in the area that the IP normally services. If the IP 23
does not store the particular template or feature information it
needs to process a call, the IP 23 can communicate with a remote IP
23.sub.R to obtain that information. In the present shared line
example, the IP 23 will store template or feature data for each
subscriber associated with the particular off-hook line.
When the IP 23 receives input speech and extracts the
characteristic information during actual call processing, the IP
compares the extracted speech information to stored pattern
information, to identity and authenticate the particular caller. In
the present example, the voice authentication module 233 in the IP
23 compares the extracted speech information to the stored template
or feature data for each subscriber associated with the particular
off-hook line. This includes the children A and B.
The IP 23 determines if the information extracted from the speech
input matches any of the stored template data feature data for an
identifiable subscriber. If there is a match, the IP now knows the
identity of the calling subscriber. Based on the identification of
the calling subscriber, the IP 23 selects a virtual office
equipment (OE) number from storage that corresponds to the
subscriber.
The IP 23 formulates a D-channel signaling message containing the
virtual office equipment (OE) number together with an instruction
to load that OE number into the register assigned to the call in
place of the OE number of the off-hook line. The IP 23 supplies the
message to the SSP central office switch 11.sub.1 over the
D-channel of the ISDN PRI link. In response, the administrative
module processor 61 rewrites the OE number in the register assigned
to the call using the OE number received from the IP 23.
Upon rewriting the OE number in the register, the administrative
module processor 61 of central office switch 11.sub.1 also reloads
the profile information in the register. Specifically, the
administrative module processor 61 retrieves profile information
associated with the virtual office equipment (OE) number from the
disc storage 63 into the register. As such, the profile information
in the assigned register in the call store 67 now corresponds to
the identified subscriber, rather than to the off-hook line.
The profile information provides a wide range of data relating to
the subscriber's services, including the permissions and
restrictions applicable to the involved children. In the presently
preferred implementation, when the central office switch 11.sub.1
reloads the profile, the central office disconnects the link to the
IP 23 and connects tone receivers to the caller's line. Optionally,
the central office 11.sub.1 may provide a `dial tone` or other
message over the line. The caller now dials digits in the normal
manner, and the switch in the central office 11.sub.1 loads the
dialed digits into the assigned register within the call store 67.
The central office 11 utilizes the dialed digits and the
subscriber's profile data to process the call. If the dialed digits
represent a call permitted to the caller further processing
proceeds. On the other hand, if the number is not included in those
which are permitted to the particular caller one of several
alternative steps may follow. In the simplest situation the call
processing may be discontinued with or without an audio
announcement to the caller. As an alternative the call may be
completed to a directory number supplied by the parent or guardian
who then admonishes the child.
Assuming that the dialed digits match digits stored in the callers
profile, it is a feature of the invention that actual verification
of the authenticity of the responding party is performed. To this
end an IAM message is sent to the destination SSP containing data
in addition to that which is typically carried. This data
information instructs the SSP to execute a pre-designated
verification procedure. According to one preferred procedure the
destination SSP sets up a voice connection between the IP and the
called terminal. This is established via data signaling similar to
that described in establishing a voice link between the originating
central office and the IP for the originating end voice
processing.
The availability of the called terminal is established by standard
CCIS signaling, ringing signals are sent, and a responding party
goes off-hook. Again a challenge prompt is delivered requesting the
name of the responding party. When this is provided the signal is
processed in the IP against a pre-prepared template which is
mandated by the personal profile of the caller. Assuming a match is
established, this is signaled by the IP to the originating switch
and a trunk connection is established between the calling and
called terminals. If no match can be established after a
pre-specified number of attempts the caller is advised and the call
processing discontinued. An audio announcement to the calling party
is preferably provided.
According to yet another feature of the invention a system is
provided for protecting the subscriber against the calls of
stalkers or other recurring threatening calls. In this situation it
is assumed that the unwanted calls have been received a sufficient
number of times to allow the called line to record and create voice
templates for the threatening caller. These may include the name or
pseudonym used by the caller and optionally the name of the called
party, where the stalker is calling for a particular person.
In the handling of this type of call pursuant to one preferred
embodiment of the invention, the protected or guarded line and
directory number have a terminating attempt trigger (TAT) set
against the particular line. When the central office SSP which
serves that line or local loop detects a call (receives the IAM) to
that line, it loads profile information associated with the called
line. The loaded profile data indicates a terminating attempt
trigger (TAT) set against the particular line. The SSP interacts
with the SCP, and finds that identification of the called party is
necessary. In accord with directions from the SCP the call is
routed to the IP.
The IP prompts the caller to identify a desired called party. The
IP uses speech recognition processing to identify the desired
called party from those associated with the premises line. Based on
identification of the called subscriber, the IP signals the SSP
switch to substitute the CPR or profile of the now identified
called party for the presently loaded CPR designated by the line
OE. Thus the virtual OE profile of the specific called party is
substituted for the line profile.
This substitution having been accomplished, the installed profile
contains data information which indicates that the identity of the
calling party is to be sought. The IP is apprised of this
requirement and uses another prompt to the calling party to
identify himself. This may be a prompt such as "Who is calling?". A
template for the name of the harassing party is preferably
maintained in the IP. This template is now used by the IP to verify
that the caller is in fact the harasser or stalker. Alternatively
the IP speech recognition module is trained to recognize the
name.
The virtual OE profile of the called party contains data
information for further handling of the call. A number of
alternatives may be provided either singly or in combination. The
profile may direct that the serving central office forward the call
to a specified directory number of a third station. This station
may constitute a terminal of a police authority or investigative
organization. Police pursuit of the caller may ensue if sufficient
information is available. The terminal may also record the ensuing
dialog. In the case where a stalker situation pertains to minors,
the call may be forwarded to a station of a parent or guardian.
This option may occur either as an alternative or in addition to
forwarding the call to the station of a police authority. Again
recording may be performed if appropriate.
While the foregoing handling of an undesired call has been
described in terms of a stalker, it will be understood that the
methodology may be applicable to other types of calls. As a further
example, pornographic calls may be intercepted and handled in a
like manner. Assuming the satisfaction of applicable legal
requirements evidence may be gathered to assist in criminal or
civil legal proceedings.
In addition to the foregoing, the invention also comprehends
providing assistance to authorities and police in the apprehension
of wanted individuals. To this end the instant embodiment of the
invention includes identification of the site of origination of the
offending call. This information may then be automatically brought
to the attention of the cognizant law enforcement authority on a
real time basis. As a further alternative the information may be
used to trigger other legally sanctioned investigatory or police
action.
In the type of two party situation now under consideration the
calling target individual may be subject to a court order
prohibiting contact with the called party. In addition to the
prohibition, the court order may authorize monitoring and recording
of any prohibited calls and apprehension of the offending party.
Such situations may be encountered with stalkers, pornographic
callers, in troubled marital situations, among others. In the cases
of stalkers and pornographic callers there are two party situations
involving illegal activity, although the caller may be identified
only by voice. However, monitoring and recording may be authorized
as well as apprehension of the offender.
In the known stalker situation there may be an outstanding order
for arrest which is not limited to apprehending the stalker when
engaged in the illegal activity. However, the stalker may be known
only by voice identification and the whereabouts of the stalker is
usually unknown. Under such circumstances the methodology of this
embodiment of the invention provides an opportunity to track the
location of the stalker and to possibly gain further information to
permit more specific identification. Identification of the site of
the stalker or other such caller during actual engagement in a
prohibited telephone conversation may offer the optimal possibility
of effecting prompt apprehension. It will be appreciated that even
though there may be an outstanding order for the arrest of the
person involved, it is impractical, as well as violative of legal
rights, to indiscriminately monitor a large number of telephone
lines of uninvolved and uninformed parties. The two party situation
offers both a legal and practical application for the use of the
features of the present embodiment of the invention.
In one illustrative application of this embodiment of the
invention, the aggrieved individual and associated subscriber
terminal are known. This information identifies the subscriber line
and its office equipment or OE number or designation, the aggrieved
individual, and his or her virtual OE number. It is assumed that
the speech processing facility, preferably an intelligent
peripheral or IP has been provided with appropriate speech
templates to implement speech identification and/or authentication
of an offending voice identifiable individual. The virtual OE
designation of the aggrieved individual identifies a personal
customer profile record (CPR) for the aggrieved party. This profile
defines procedures to be followed upon the aggrieved party
receiving a call from a caller who has been previously identified
by his or her speech as the wanted offending party.
An illustrative example of the operation of this embodiment of the
invention under these circumstances is now described. In this case
speech recognition is used with respect to the voice of the calling
party. Current speech recognition technology permits recognition
with a reasonable degree of certitude based on training from a
limited sample of recorded speech of a subject. In a situation of
this type the target of the speech recognition is a wanted person
and is not likely to cooperatively participate in any type of
prompting procedure which may seem suspicious. As a result, it may
be necessary to rely on such sophisticated speech recognition
techniques as applied to random speech. On the other hand, in some
instances it may be found possible to obtain speech templates of
the target person uttering the name of the aggrieved party. This
alone may prove sufficient to provide the necessary
identification.
In such situations the instant embodiment of the invention relies
on the speech recognition capability of the module 235 in the IP
23. The speech recognition module 235 enables the IP to analyze
incoming audio information to recognize vocabulary words. The IP
interprets the spoken words and phrases to determine subsequent
action. For example, the IP may recognize the target caller
speaking the name of a called subscriber and use the subscriber
identification obtained in this manner to instruct the terminating
central office to thereafter control the call in accord with that
named subscriber's profile. On the other hand, the target caller
may simply go on hook if he or she does not recognize the voice of
the aggrieved party answering the telephone, so that another speech
recognition procedure is necessary.
In this example the offending or target party may go off hook at
any telephone in the relevant network. This creates a corresponding
signal or change in state on the line to the central office to
which the telephone is connected. In the call sequence, the
off-hook signal acts as a type of service request, i.e. a request
to make an outgoing call. The originating central office detects
the off-hook and commences its call processing. Specifically, the
originating central office assigns a register to the call in the
call store of the originating central office switch. The switch
loads profile information associated with the off-hook line from
the disc storage of the switch into the assigned register.
The originating central office provides dial tone or the like over
the line, the caller dials digits corresponding to the desired
destination and the switch in the originating central office begins
its processing to route the call through the network. The
originating central office uses the dialed number to initiate a
CCIS communication with the exchange serving the intended
destination, in this example the terminating central office, which
is assumed to be an SSP central office.
Specifically, the originating central office generates an Initial
Address Message (IAM) for transmission to the terminating central
office. The IAM message includes the SS7 destination point code
(DPC) of the terminating central office and the SS7 origination
point code (OPC) of the originating central office for addressing
purposes. The payload portion of the IAM message includes the
called and calling numbers. The originating central office
transmits the IAM message through the CCIS network to the distant
terminating office.
When the terminating office receives the IAM message, the
administrative module processor for that office retrieves the
customer profile for the number in the destination number field of
that message (e.g. the number for the telephone line identified by
OE in the destination central office for that number), from its
mass storage system and loads that profile into one of its call
store registers.
The subscriber for that line has personal dial tone service and a
virtual OE assignment for each individual associated with that
service. Usually such individuals reside at the site or subscriber
premises at which the line or local loop terminates. The loaded
profile for the OE of the line itself indicates a terminating
trigger for that line and OE. The office of the subscriber, being
an SSP type office, detects this call PIC as an AIN trigger.
In response to the IAM and the terminating attempt trigger (TAT)
set in the subscriber's profile, the SSP type terminating central
office switch launches a query to the SCP. Specifically, the SSP
creates a TCAP query message containing relevant information, such
as the office equipment (OE) number assigned to the called number
line, and transmits that query over an SS7 link to one of the STPs.
The query includes a destination point code and/or a global title
translation addressing the message to the SCP, and the STP relays
the query message over the appropriate link to the SCP. The query
from the SSP terminating central office identifies the called line
by its associated office equipment (OE) number and possibly by a
single telephone number associated with the called line.
In response to a query, the SCP accesses its a database, typically,
the MSAP database set up in the ISCP, to determine how to process
the particular call. The SCP identifies an access key in the query
and uses the key to retrieve the appropriate record from the
database. In this case, the query indicates a terminating attempt
trigger as the trigger event, therefore the SCP uses the called
line office equipment (OE) number as the access key. The SCP
retrieves a call processing record (CPR) corresponding to the
office equipment (OE) number associated with the called line and
proceeds in accord with that CPR.
The call processing by the destination central office switch
utilizes the loaded subscriber profile information. In this
instance the called station subscribes to personal dial tone
service and the line OE is associated with multiple virtual OEs
assigned to individuals residing at the same subscriber premises.
Thus the subscriber profile of the OE specifies procedures to be
followed to implement the particular personal dial tone service
desired by the subscriber.
In the instant preferred embodiment of the invention a situation
exists wherein an individual to whom one of the virtual OEs is
assigned is being harassed or threatened by an individual who has
been identified by voice only. The full identity and location of
that target individual is not known. However, the threatened
individual has been repeatedly harassed by telephone at the
subscriber premises via the subscriber line. It is desired to
locate and apprehend the offending party.
In order that this may be accomplished according to this preferred
embodiment of the invention, it is desired to have the offender
making a call to the threatened party speak a specific or
sufficiently extended utterance to permit the calling party to be
identified as the offending individual. According to one procedure,
the CPR of the subscriber line OE for the premises may contain
instructions to deliver a prompt to all callers. That prompt could
request all such callers to speak the name of the called party. In
this procedure the IP is equipped with speech templates to permit
speech identification/verification or SIV of a specific virtual OE
from the utterance of the name by the caller. However, if the
caller is the offending party, he or she is likely to be wary and
may be suspicious of such a prompt delivered request. In that event
the party may simply go on-hook and thwart any possibility of
identification or apprehension.
A more preferred procedure, pursuant to one feature of this
embodiment of the invention, is to instruct all residents at the
subscriber premises to always answer the telephone by speaking
their own name, such as, "This is Jane" or "This is John," as the
case may be. The IP has been provided with voice trained templates
to enable identification of the desired virtual OE of each named
individual from such a name utterance.
In response to the IAM, and to the terminating attempt trigger
(TAT) set in the subscriber's profile, and to the procedures
specified in the subscriber's profile, the SSP switch routes the
call to the nearest IP having the necessary SIV capability. A two
way voice grade call connection now extends between the called line
and the IP. As noted above, the communication link to the IP 23
provides both line connections and signaling, preferably over a
primary rate interface (PRI) type ISDN link. When the central
office 11.sub.1 extends the call from the calling party's line to a
line circuit (over a B channel) to the IP 23, the switch in that
office also provides call related data over the signaling link (D
channel for ISDN). The destination central office switch now
provides ringing signal to the called line via its OE. As a result
one of the subscribers at the premises goes off-hook. It is
optional as to whether or not the calling party is voice connected
at this stage.
The next action is dependent on the identity of the person at the
subscriber premises who answers the telephone. In this example it
is assumed that the threatened or harassed individual answers. As
previously stated, all answering parties are instructed to first
identify themselves. Here the threatened answering party does so,
as for example, "This is Jane." When the IP receives input speech
and extracts the characteristic information during actual call
processing, it compares the extracted speech information to stored
pattern information to identity and authenticate the particular
answering party or subscriber. In the present example, the voice
authentication module 233 in the IP 23 compares the extracted
speech information to the stored template or feature data for each
subscriber associated with the particular off-hook line. The IP now
knows the identity of the called subscriber. Based on the
identification of the called subscriber, the IP 23 selects a
virtual office equipment (OE) number from storage that corresponds
to the subscriber.
The IP 23 formulates a D-channel signaling message containing the
virtual office equipment (OE) number together with an instruction
to load that OE number into the register assigned to the call in
place of the OE number of the off-hook line. The IP 23 supplies the
message to the SSP central office switch 11.sub.1 over the
D-channel of the ISDN PRI link (step S14). In response, the
administrative module processor 61 rewrites the OE number in the
register assigned to the call using the OE number received from the
IP 23.
Upon rewriting the OE number in the register, the administrative
module processor 61 of central office switch 11.sub.1 also reloads
the profile information in the register. Specifically, the
administrative module processor 61 retrieves profile information
associated with the virtual office equipment (OE) number from the
disc storage 63 into the register. As such, the profile information
in the assigned register in the call store 67 now corresponds to
the identified subscriber, rather than to the off-hook line.
The profile information provides a wide range of data relating to
the subscriber's services. Included in that information the profile
data provides necessary instructions to alert the IP to prepare to
attempt to match the speech of the calling party to identify the
calling party as the target. If the procedure which is specified in
the CPR includes the option of not voice connecting the calling
party to the called line to hear the live response of the answering
party subscriber, the IP is directed to record the response for
playback to the calling party when that party has been voice
connected. The calling party is now connected to the called line,
which has a voice connection to the IP, i.e., it is bridged to the
IP. The IP plays the recorded response to the calling party and
prepares to monitor the speech of the caller and attempt to match
it to that of the target.
If the IP is able to establish a match through its SIV procedures,
the conversation is recorded. In addition, and as an example of
procedures which may be specified according to one preferred
embodiment of the invention, the loaded profile directs that the
destination SSP interact with the IP to use the identity of the
calling telephone number to identify the site of the calling
telephone, and to send that information to the police. It is
assumed that the police have been previously alerted to the
situation and have authority and orders to apprehend the offending
individual. As an attempt to provide the police with maximum time
to respond, the threatened individual may have been requested to
attempt to hold the offending party on the line as long as
possible. The IP may be maintained in a bridged condition in order
to monitor and record the conversation as a legal evidence
procurement measure. In addition the conversation may also be
bridged to the police.
The forwarding of the information locating the calling telephone
may be implemented as follows. The loaded profile of the harassed
party includes data that directs and initiates a sequence such as
the following. The originating end office has addressed and
transmitted the IAM message with the called and calling telephone
numbers. The terminating end office launches a second query message
through one or more of the STP(s) to the LIDB database 21 in FIG.
1. The query message includes both the telephone number associated
with the calling station or its telephone line as well as a code
identifying the virtual OE and nature of this request.
The LIDB database uses the calling party telephone number and the
code received in the query, to retrieve that calling subscriber
terminal or line account file record from the database. This
includes the name and address of the subscriber for that telephone
station. The LIDB database 21 compiles a TCAP call control message
including the name and address data and returns that call control
message to the terminating central office via the SS7 network. The
terminating central office switching system receives the call
control message from the LIDB database and shares this information
with the IP via the signaling circuit. The same information is
transmitted to the police via signaling and/or voice circuit. The
police are now bridged onto the voice conversation and have been
provided with the address from which the offending call is being
conducted. If a patrol car is in the vicinity of the identified
address, or if the harassed person is able to hold the offending
caller on the line a sufficient time, the car may be radio
dispatched and an apprehension may be accomplished.
In the foregoing example it was assumed that the harassed party
answered the telephone. The profile of that person was then
installed based on that person identifying his or herself in
answering the telephone. In the case where the telephone is
answered by a resident of the premises other than the harassed
party, that party answers with the same pre-specified greeting,
such as, "This is John." In this situation the profile in the CPR
corresponding to the subscriber line may specify that the profile
of that individual, namely John, be substituted for the line CPR.
Unless the caller specifically requests to speak to the harassed
party by name, the call will proceed in accord with the CPR of the
answering party, in this instance, John.
If the caller, as yet unidentified, asks to speak to the harassed
party, Jane in this example, alternative procedures may be utilized
according to this embodiment of the invention. According to a first
procedure, the answering party, John, calls the harassed party,
Jane, in such a manner that his call for Jane is fully audible to
the telephone microphone. This utterance is identified by the
bridged IP SIV, and the CPR of Jane is substituted by the
destination central office switch. Jane answers in the pre-agreed
format, such as, "This is Jane." The CPR for Jane has been entered
and the same procedure is followed as has been previously described
in the instance in which Jane answered the telephone.
As an alternative to this procedure, an answering party other than
Jane may say "Please hold," place the call on hold, and call Jane
to the telephone. Jane may then remove the call from hold, and
answer in the pre-agreed manner, such as, "This is Jane." The CPRs
of all residents of the subscriber premises may contain call
handling instruction data directing interpretation of hold signals
on this line as directing connection to the SIV facilities of the
IP and directing the IP to stand by to implement CPR selection
corresponding to the next matching name of a subscriber. As a
result the switch substitutes the CPR for Jane on identifying her
name through SIV. The CPR for Jane is now entered and the same
procedure is followed as has been previously described in the
instance in which Jane answered the telephone. As a still further
alternative to the foregoing, all subscriber OE profiles at the
subscriber premises may contain processing instructions to cause
the IP to be connected upon execution of a *HOLD sequence. In this
case all subscribers are instructed to use a *HOLD sequence when
calling the threatened party to the telephone.
Reference is now had to FIG. 6 to facilitate description of yet
another embodiment of the inventions. The present day popularity of
voice mail service has been previously mentioned in discussing the
prior art background of the present inventions. There has also been
mention of the Intelligent Peripheral or IP 23 (FIGS. 1 and 3)
having a voice mail server 239 for use by the network illustrated
in FIG. 1. FIG. 6 provides a diagrammatic depiction of one
available voice mail server suited for use in a preferred
embodiment of an implementation for providing voice mail services
in the present multi-subscriber per line environment.
Referring to FIG. 6, the centralized message service or voice mail
system in the illustrated example comprises voice messaging
equipment such as a voice mail system 120. The voice mail system
120 includes a digital switching system (DSS) 121, a master control
unit (MCU) 123, a number of voice processing units (VPUs) 125 and a
master interface unit (MIU) or concentrator 127. The master control
unit (MCU) 123 of the voice mail system 120 is a personal computer
type device programmed to control overall operations of the system
120.
Each of the voice processing units 125 also is a personal computer
type device. The voice processing units 125 each include or connect
to one or more digital mass storage type memory units (not shown)
in which the actual messages are stored. The mass storage units,
for example, may comprise magnetic disc type memory devices.
Although not specifically illustrated in the drawing, the voice
processing units 125 also include appropriate circuitry to transmit
and receive audio signals via T1 type digital audio lines. An
ETHERNET type digital network 129 carries data signals between the
MCU 123 and the voice processing units 125. The Ethernet network
129 also carries stored messages, in digital data form, between the
various voice processing units 125. The system 120 further includes
T1 type digitized audio links 128 between the DSS switch 121 and
each of the voice processing units 125.
The voice mail system 120 connects to the central office switching
system 11 via the network 240 and the direct talk modules 231A and
231B and ISDN PRI TRUNKS which provide voice and signaling
channels. Communication with the SSPs in the central offices may
also be had via the network 240, IP communication server 243 and
router 241. The MIU 127 is a data concentrator which effectively
provides a single connection of many data signal links into the MCU
123 of the voice mail system.
The above described voice mail system architecture is similar to
existing voice mail type central messaging systems, such as
disclosed in U.S. Pat. No. 5,029,199 to Jones et al., although
other messaging system architectures such as disclosed in the other
patents cited above could be used. See also U.S. Pat. No. 5,661,782
to Farris and Bartholomew for additional description of operation
of this type of voice mail system.
For each party who subscribes to a voice mail service provided by
the centralized messaging system 120, the MCU 123 stores
information designating one of the voice processing units 125 as
the "home" unit for that subscriber. Each voice processing unit 125
stores generic elements of prompt messages in a common area of its
memory. Personalized elements of prompt messages, for example
recorded representations of each subscriber's name spoken in the
subscriber's own voice, are stored in designated memory locations
within the subscriber's "home" voice processing unit.
In voice mail systems of the type discussed above, a subscriber's
"mailbox" does not actually correspond to a particular area of
memory. Instead, the messages are stored in each "mailbox" by
storing appropriate identification or tag data to identify the
subscriber or subscriber's mailbox to which each message
corresponds.
Each time a call comes in to the voice mail system 120, the master
control unit 123 controls the digital switching system 121 to
provide a multiplexed voice channel connection through to one of
the voice processing units 125. Typically, the call connection goes
to the "home" voice processing unit for the relevant subscriber.
The voice mail subscriber is identified by data transmitted from
the switching system 11, as described above, if the call is a
forwarded call. If all 24 T1 channels to the "home" voice
processing unit are engaged, the central processing unit 123
controls switch 121 to route the call to another voice processing
unit 125 which is currently available.
The voice processing unit connected to the call retrieves prompt
messages and/or previously stored messages from its memory and
transmits them back to the calling party via the internal T1 line
128, the DSS switch 121 one of the voice channels, central office
switching system 11 and the calling party's telephone line. The
voice processing unit 125 connected to the call receives incoming
messages from the caller through a similar route and stores those
messages in digital form in its associated mass storage device.
When the incoming call is a forwarded call, the connected voice
processing unit 125 provides an answering prompt message to the
caller, typically including a personalized message recorded by the
called subscriber. After the prompt, the voice processing unit 125
records a message from the caller and identifies that stored
message as one for the called subscriber's mailbox.
At times the connected voice processing unit 125 will not have all
necessary outgoing messages stored within its own associated
memory. For example, a forwarded call normally will be connected to
the called subscriber's "home" voice processing unit 125, but if
the home unit is not available the forwarded call will be connected
to a voice processing unit 125 other than the subscriber's home
voice processing unit. In such a case, the connected unit 125
requests and receives from the home unit 125 the personalized
components of the answering prompt message via the data network
129. The connected voice processing unit 125 will store the
transferred message data in its own memory, and when necessary,
will play back the transferred data from its own memory as outgoing
messages in the exact same manner as for any prompts or greeting
messages originally stored in its own memory.
The connected voice processing unit 125 also will store any
incoming message in its own associated memory together with data
identifying the message as one stored for the called subscriber's
mailbox. As a result, the system 120 actually may store a number of
messages for any given subscriber or mailbox in several different
voice processing units 125. Subsequently, when the voice mail
subscriber calls in to the voice mail system 120 to access the
subscriber's mailbox, the call is connected to one voice processing
unit 125. Again, this call typically goes to the home unit 125 but
would go to a different available one of the units 125 if the home
unit is not available at the time. In response to appropriate DTMF
control signals, or preferably voice signals, received from the
subscriber, the connected voice processing unit retrieves the
subscriber's messages from its own memory and plays the messages
back to the subscriber. If any messages are stored in other voice
processing units, the connected unit 125 sends a request the other
units 125 to download any messages for the subscriber's mailbox
those units have actually stored. The downloaded messages are
stored in the memory of the connected voice processing unit 125
which replays them to the subscriber.
In a typical usage of the present embodiment of the invention there
may be four subscribers for the telephone station 1.sub.A which is
connected to a line or local loop having a single telephone number
and office equipment or OE number. Each of the four subscribers is
provided with a personalized service profile or customer profile
(CPR) which is identified by a virtual office equipment or OE
number. As previously explained a virtual office equipment number
refers to "virtual" equipment which has no real existence in the
relevant central office. The profiles include for each subscriber a
range of information relating to subscribers services, such as
service features, classes of service, individual billing options,
information relating to restrictions applied to individual users,
as well as the performance of functions related to that user. Each
of the four subscribers in this example subscribes to voice mail
service and is assigned a mail box or partition of a mail box in
the voice mail server 239 in the intelligent peripheral or IP 23.
It will be understood that while the voice mail unit or facility is
here shown as integrated into the IP this need not be the case. The
voice mail unit may be provided as a stand alone unit as shown, for
example in the above cited Jones et al. Patent. The subscription to
the voice mail service and identification of a mail box is included
in the information in each customer profile relating to the
specific subscriber. Similarly the information data may prescribe
that a call is to be forwarded to the voice mail system on a `busy`
and/or `no-answer` condition.
When the serving central office SSP 11 detects a call to a line
having the personalized service, processing hits a terminating
attempt trigger (TAT). The SSP interacts with the SCP 19 and routes
the call to the IP 23. The IP 23 prompts the caller to identify a
desired called party, e.g. one of the subscribers sharing the
single line. Menu announcement together with speech or voice
utterance recognition processing by the IP 23 enables
identification of the desired called party from those subscribers
associated with the line. Based on identification of the called
subscriber, the IP 23 signals the SSP switch 11 to load profile
data for that subscriber into the register assigned to the call in
the call store. The switch 11 thereupon uses the selectively loaded
personal profile information for terminating the call. The IP
disconnects, and the SSP central office 11 processes the call in
accord with the loaded profile information which is identified by
the virtual OE of the called and now identified subscriber. As an
alternative or conjunctively with this procedure, the SSP may first
load into the assigned register a generic customer profile for the
OE for the line itself. Such a service profile may contain
information which is generic to the subscribers served by that line
and OE umber, particularly subscribers who may not subscribe to any
enhanced services.
For example, in accord with the now loaded profile the central
office 11 may provide a distinctive ringing signal corresponding to
the identified subscriber. This service enables distinctive ringing
for multiple subscribers on one line without assigning each
subscriber a separate telephone number. In this example the loaded
profile information specifies forwarding of the call to the
identified subscriber's identified mailbox in event of a busy or
no-answer condition.
It is a feature of this embodiment of the invention that the call
is initially routed to the intelligent peripheral. The IP includes
a so-called `challenge` wherein the caller is requested to speak
his or her name. The intelligent peripheral speech identification
capability includes a previously obtained template or templates to
permit identification of the called party through use of that
template or equivalent speech identification/verification
procedure. Upon making the identification the IP selects a virtual
office equipment (OE) number from the storage that corresponds to
the subscriber. The IP then sends instructions to load that OE
number into the register assigned to the call in place of the OE
number of the subscriber premises line or local loop. In response,
the administrative module processor 61 rewrites the OE number in
the register assigned to the call using the OE number received from
the IP 23.
Upon rewriting the OE number in the register, the administrative
module processor 61 of central office switch 11.sub.1 also reloads
the profile information in the register. Specifically, the
administrative module processor 61 retrieves profile information
associated with the virtual office equipment (OE) number from the
disc storage 63 into the register. As such, the profile information
in the assigned register in the call store 67 now corresponds to
the identified subscriber, rather than to the customer premises
line. Having loaded the proper customer profile record in the
assigned call store register the destination central office
utilizes the dialed digits and the now loaded subscriber's profile
data to process the call. Thus, the generic profile corresponding
to the OE number (`real OE number` for actual equipment) may be
replaced by the profile corresponding to a virtual OE number.
Pursuant to the loaded profile the central office transmits to the
called premises line a distinctive ringing signal which identifies
the subscriber corresponding to the virtual OE number and profile.
Also pursuant to the profile a no answer (or busy/no answer)
condition results in forwarding the call to the intelligent
peripheral via the voice and signaling links. When the call comes
in to the voice mail system, the master control unit 123 controls
the digital switching system 121 to provide a multiplexed voice
channel connection through to one of the voice processing units
125. Typically, the call connection goes to the "home" voice
processing unit for the relevant subscriber. The voice mail
subscriber is identified by data transmitted from the switching
system 11, as described above.
The voice processing unit connected to the call retrieves prompt
messages and/or previously stored messages from its memory and
transmits them back to the calling party via the internal T1 line
128, the DSS switch 121 one of the voice channels, central office
switching system 11 and the calling party's telephone line. The
voice processing unit 125 connected to the call receives incoming
messages from the caller through a similar route and stores those
messages in digital form in its associated mass storage device.
The connected voice processing unit 125 provides an answering
prompt message to the caller, typically including a personalized
message recorded by the called subscriber. After the prompt, the
voice processing unit 125 records a message from the caller and
identifies that stored message as one for the called subscriber's
mailbox. Subsequently, when the voice mail subscriber calls in to
the voice mail system 120 to access the subscriber's mailbox, the
call is connected to one voice processing unit 125. Again, this
call typically goes to the home unit 125 but would go to a
different available one of the units 125 if the home unit is not
available at the time. In response to appropriate DTMF control
signals received from the subscriber, the connected voice
processing unit retrieves the subscriber's messages from its own
memory and plays the messages back to the subscriber. If any
messages are stored in other voice processing units, the connected
unit 125 sends a request the other units 125 to download any
messages for the subscriber's mailbox those units have actually
stored. The downloaded messages are stored in the memory of the
connected voice processing unit 125 which replays them to the
subscriber.
According to another feature of this embodiment of the invention
the subscriber's profile contains data which prescribes a
distinctive interrupted dial tone to indicate the presence of
stored voice mail. Each of the four subscribers in this example
subscribes to voice mail service and is assigned a mail box or
partition of a mail box in the voice mail server 239 in the
intelligent peripheral or IP 23. Each subscriber may also be
assigned a distinctive interrupted dial tone, such as, for example,
two short tones, a long and a short, etc., to indicate the presence
of voice mail. However, such a distinctive tone is not necessary as
will now become apparent.
As has been previously described, when one of the subscribers in
the relevant multi-subscriber per line premises goes off-hook, this
is interpreted as a request to make an outgoing call. The
associated central office commences its call processing.
Specifically, the central office assigns a register in the call
store 67 to this call and loads profile information associated with
the off-hook line from the disc storage 63 into the assigned
register. In this case, the central office 11.sub.1 is an SSP
capable office, and the loaded profile data indicates an off-hook
immediate trigger set against the particular line. The serving SSP
type office 11.sub.1 therefore detects this off-hook PIC as an AIN
trigger.
In response to the off-hook and the off-hook trigger set in the
subscriber's profile, the SSP type central office switch 11.sub.1
launches a query to the SCP 19. Specifically, the SSP 11.sub.1
creates a TCAP query message containing relevant information, such
as the office equipment (OE) number assigned to the off-hook line,
and transmits that query over an SS7 link to one of the STPs 15.
The STP relays the query message over the appropriate link to the
SCP 19. The query from the SSP central office 11.sub.1 identifies
the caller's line by its associated office equipment (OE) number
and possibly by a single telephone number associated with the
off-hook line.
In response to a query, the SCP 19 accesses its a database,
typically, the MSAP database set up in the ISCP, to determine how
to process the particular call. The SCP 19 identifies an access key
in the query and uses the key to retrieve the appropriate record
from the database. In this case, the query indicates an off-hook
trigger as the trigger event, therefore the SCP 19 uses the calling
party office equipment (OE) number as the access key. The SCP 19
retrieves a call processing record (CPR) corresponding to the
office equipment (OE) number associated with the off-hook line and
proceeds in accord with that CPR.
The CPR provides the information necessary for routing the call to
a node of the network that will perform speaker
identification/verification (SIV), in this example the Intelligent
Peripheral or IP. Based on the CPR, the SCP 19 formulates a
response message instructing the SSP central office 11.sub.1
serving the customer to route the call. In this case, the message
includes an office equipment (OE) number or telephone number, used
for routing a call to the identified IP 23. The SCP 19 formulates a
TCAP message in SS7 format, with the destination point code
identifying the SSP office 11.sub.1. The SCP 19 transmits the TCAP
response message back over the SS7 link to the STP 15, and the STP
15 in turn routes the TCAP message to the SSP central office
11.sub.1.
The SSP type switch in the central office 11.sub.1 uses the routing
information to connect the call to one of the lines or channels to
the IP 23. A two-way voice grade call connection now extends
between the calling station 1.sub.A and the IP 23. In the present
example, the switch actually connects the off-hook line to the line
to the IP before providing dial tone.
As noted above, the communication link to the IP 23 provides both
line connections and signaling, preferably over a primary rate
interface (PRI) type ISDN link. When the central office 11.sub.1
extends the call from the calling party's line to a line circuit
(over a B channel) to the IP 23, the switch in that office also
provides call related data over the signaling link (D channel for
ISDN). The call related data, for example, includes the office
equipment (OE) number normally associated with the off-hook line
and possibly the telephone number for that line.
In response to the incoming call, the IP 23 seizes the line, and
launches its own query to the SCP 19. In the preferred network
illustrated in FIG. 1, the IP 23 and the SCP 19 communicate with
each other via the separate second signaling network 27. The query
from the IP 23 again identifies the caller's line by at least its
associated office equipment (OE) number.
In response to the query from the IP 23, the SCP 19 again accesses
the appropriate CPR and provides a responsive instruction back
through the network 27 to the IP 23. The IP issues a `Challenge
Phase` to prompt the user to input specific identifying
information. In this case, the instruction causes the IP 23 to
provide a prompt message over the connection to the caller. Here,
the instruction from the SCP 19 causes the IP 23 to provide an
audio announcement prompting the caller to speak personal
information. In one preferred example, the IP plays an audio prompt
message asking the caller, `Please say your full name`. As
previously explained the process may request any appropriate
identifying information.
The signal received by the IP 23 goes over the lines and through
the central office switch(es) for presentation via the off-hook
telephone 1.sub.A to the calling party. In response, the caller
speaks identifying information into their off-hook telephone, and
the network transports the audio signal to the IP 23.
As noted above, an IP 23 can provide a wide range of call
processing functions. In this example, the IP performs speaker
identification/verification (SIV) on the audio signal received from
the off-hook telephone. When the IP 23 receives speech input
information during actual call processing, for this service
example, the IP analyzes the speech to extract certain
characteristic information. As has been described above the IP
stores templates or feature data for each subscriber associated
with the particular off-hook line.
When the IP 23 receives input speech and extracts the
characteristic information during actual call processing, the IP
compares the extracted speech information to stored pattern
information, to identity and authenticate the particular caller. In
the present example, the voice authentication module 233 in the IP
23 compares the extracted speech information to the stored template
or feature data for each subscriber associated with the particular
off-hook line.
The IP 23 determines if the information extracted from the speech
input matches any of the stored template data feature data for an
identifiable subscriber. If there is a match, the IP now knows the
identity of the subscriber who went off-hook. Based on the
identification of the calling subscriber, the IP 23 selects a
virtual office equipment (OE) number from storage that corresponds
to the subscriber.
The IP 23 formulates a D-channel signaling message containing the
virtual office equipment (OE) number together with an instruction
to load that OE number into the register assigned to the call in
place of the OE number of the off-hook line. The IP 23 supplies the
message to the SSP central office switch 11.sub.1 over the
D-channel of the ISDN PRI link. In response, the administrative
module processor 61 rewrites the OE number in the register assigned
to the call using the OE number received from the IP 23.
Upon rewriting the OE number in the register, the administrative
module processor 61 of central office switch 11.sub.1 also reloads
the profile information in the register. Specifically, the
administrative module processor 61 retrieves profile information
associated with the virtual office equipment (OE) number from the
disc storage 63 into the register. As such, the profile information
in the assigned register in the call store 67 now corresponds to
the identified subscriber, rather than to the off-hook line.
As described, the profile information provides a wide range of data
relating to the subscriber's services. Included in this information
is identification of this subscriber's mailbox and instructions for
returning to the off-hook party a dial tone indicative of the
presence of unread voice mail in that party's mailbox. This signal
is delivered to the party who can then retrieve the message in the
usual manner.
As a further feature of this embodiment of the invention, the
subscriber profile may also contain instructions to permit a caller
to elect to call directly to the mail box of any subscriber to the
multi-subscriber single line service. According to this feature the
SIV facility, in this instance the IP, is provided with dual
templates for each subscriber. As an example, one of the two
templates may identify the subscriber's name, whereas the other
template may identify the subscriber's name with the addition of a
command such as "mailbox." When a caller speaks the full command
"John Doe mailbox," the SIV facility interprets this as an
identification and authorization to connect the call directly to
the mailbox. In such a situation a ringing signal would be optional
when the called line is free for use. When the line is busy the
call would go directly to the IP and mailbox without the necessity
for the prompt announcing the availability of a mailbox
service.
While the foregoing has described what are considered to be
preferred embodiments of the invention, it is understood that
various modifications may be made therein and that the invention
may be implemented in various forms and embodiments, and that it
may be applied in numerous applications, only some of which have
been described herein. It is intended by the following claims to
claim all such modifications and variations which fall within the
true scope of the invention.
* * * * *